Move src/ -> webrtc/

TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org
2012-10-22 18:19:23 +00:00
parent 24a419c0c7
commit 14b43beb7c
1888 changed files with 23 additions and 23 deletions

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stefan@webrtc.org
mikhal@webrtc.org
marpan@webrtc.org
henrik.lundin@webrtc.org

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_I420_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_I420_H_
#include "video_codec_interface.h"
#include "typedefs.h"
namespace webrtc {
class I420Encoder : public VideoEncoder {
public:
I420Encoder();
virtual ~I420Encoder();
// Initialize the encoder with the information from the VideoCodec.
//
// Input:
// - codecSettings : Codec settings.
// - numberOfCores : Number of cores available for the encoder.
// - maxPayloadSize : The maximum size each payload is allowed
// to have. Usually MTU - overhead.
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK.
// <0 - Error
virtual int InitEncode(const VideoCodec* codecSettings,
int /*numberOfCores*/,
uint32_t /*maxPayloadSize*/);
// "Encode" an I420 image (as a part of a video stream). The encoded image
// will be returned to the user via the encode complete callback.
//
// Input:
// - inputImage : Image to be encoded.
// - codecSpecificInfo : Pointer to codec specific data.
// - frameType : Frame type to be sent (Key /Delta).
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK.
// <0 - Error
virtual int Encode(const VideoFrame& inputImage,
const CodecSpecificInfo* /*codecSpecificInfo*/,
const std::vector<VideoFrameType>* /*frame_types*/);
// Register an encode complete callback object.
//
// Input:
// - callback : Callback object which handles encoded images.
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
virtual int RegisterEncodeCompleteCallback(EncodedImageCallback* callback);
// Free encoder memory.
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
virtual int Release();
virtual int SetRates(uint32_t /*newBitRate*/, uint32_t /*frameRate*/)
{return WEBRTC_VIDEO_CODEC_OK;}
virtual int SetChannelParameters(uint32_t /*packetLoss*/, int /*rtt*/)
{return WEBRTC_VIDEO_CODEC_OK;}
virtual int CodecConfigParameters(uint8_t* /*buffer*/, int /*size*/)
{return WEBRTC_VIDEO_CODEC_OK;}
private:
bool _inited;
EncodedImage _encodedImage;
EncodedImageCallback* _encodedCompleteCallback;
}; // end of WebRtcI420DEncoder class
class I420Decoder : public VideoDecoder {
public:
I420Decoder();
virtual ~I420Decoder();
// Initialize the decoder.
// The user must notify the codec of width and height values.
//
// Return value : WEBRTC_VIDEO_CODEC_OK.
// <0 - Errors
virtual int InitDecode(const VideoCodec* codecSettings,
int /*numberOfCores*/);
virtual int SetCodecConfigParameters(const uint8_t* /*buffer*/, int /*size*/)
{return WEBRTC_VIDEO_CODEC_OK;};
// Decode encoded image (as a part of a video stream). The decoded image
// will be returned to the user through the decode complete callback.
//
// Input:
// - inputImage : Encoded image to be decoded
// - missingFrames : True if one or more frames have been lost
// since the previous decode call.
// - codecSpecificInfo : pointer to specific codec data
// - renderTimeMs : Render time in Ms
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK
// <0 - Error
virtual int Decode(const EncodedImage& inputImage,
bool missingFrames,
const RTPFragmentationHeader* /*fragmentation*/,
const CodecSpecificInfo* /*codecSpecificInfo*/,
int64_t /*renderTimeMs*/);
// Register a decode complete callback object.
//
// Input:
// - callback : Callback object which handles decoded images.
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
virtual int RegisterDecodeCompleteCallback(DecodedImageCallback* callback);
// Free decoder memory.
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK.
// <0 - Error
virtual int Release();
// Reset decoder state and prepare for a new call.
//
// Return value : WEBRTC_VIDEO_CODEC_OK.
// <0 - Error
virtual int Reset();
private:
VideoFrame _decodedImage;
int _width;
int _height;
bool _inited;
DecodedImageCallback* _decodeCompleteCallback;
}; // End of WebRtcI420Decoder class.
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_I420_H_

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# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
LOCAL_PATH := $(call my-dir)
include $(CLEAR_VARS)
include $(LOCAL_PATH)/../../../../../../../android-webrtc.mk
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
LOCAL_MODULE := libwebrtc_i420
LOCAL_MODULE_TAGS := optional
LOCAL_CPP_EXTENSION := .cc
LOCAL_SRC_FILES := i420.cc
# Flags passed to both C and C++ files.
LOCAL_CFLAGS := \
$(MY_WEBRTC_COMMON_DEFS)
# Include paths placed before CFLAGS/CPPFLAGS
LOCAL_C_INCLUDES := \
$(LOCAL_PATH)/../interface \
$(LOCAL_PATH)/../../../interface \
$(LOCAL_PATH)/../../../../../.. \
$(LOCAL_PATH)/../../../../../../common_video/interface \
$(LOCAL_PATH)/../../../../../../system_wrappers/interface
LOCAL_SHARED_LIBRARIES := \
libcutils \
libdl \
libstlport
ifndef NDK_ROOT
include external/stlport/libstlport.mk
endif
include $(BUILD_STATIC_LIBRARY)

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/video_coding/codecs/i420/main/interface/i420.h"
#include <string.h>
#include "common_video/libyuv/include/webrtc_libyuv.h"
namespace webrtc
{
I420Encoder::I420Encoder():
_inited(false),
_encodedImage(),
_encodedCompleteCallback(NULL)
{}
I420Encoder::~I420Encoder() {
_inited = false;
if (_encodedImage._buffer != NULL) {
delete [] _encodedImage._buffer;
_encodedImage._buffer = NULL;
}
}
int I420Encoder::Release() {
// Should allocate an encoded frame and then release it here, for that we
// actually need an init flag.
if (_encodedImage._buffer != NULL) {
delete [] _encodedImage._buffer;
_encodedImage._buffer = NULL;
}
_inited = false;
return WEBRTC_VIDEO_CODEC_OK;
}
int I420Encoder::InitEncode(const VideoCodec* codecSettings,
int /*numberOfCores*/,
uint32_t /*maxPayloadSize */) {
if (codecSettings == NULL) {
return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
}
if (codecSettings->width < 1 || codecSettings->height < 1) {
return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
}
// Allocating encoded memory.
if (_encodedImage._buffer != NULL) {
delete [] _encodedImage._buffer;
_encodedImage._buffer = NULL;
_encodedImage._size = 0;
}
const uint32_t newSize = CalcBufferSize(kI420,
codecSettings->width,
codecSettings->height);
uint8_t* newBuffer = new uint8_t[newSize];
if (newBuffer == NULL) {
return WEBRTC_VIDEO_CODEC_MEMORY;
}
_encodedImage._size = newSize;
_encodedImage._buffer = newBuffer;
// If no memory allocation, no point to init.
_inited = true;
return WEBRTC_VIDEO_CODEC_OK;
}
int I420Encoder::Encode(const VideoFrame& inputImage,
const CodecSpecificInfo* /*codecSpecificInfo*/,
const std::vector<VideoFrameType>* /*frame_types*/) {
if (!_inited) {
return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
}
if (_encodedCompleteCallback == NULL) {
return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
}
_encodedImage._frameType = kKeyFrame; // No coding.
_encodedImage._timeStamp = inputImage.TimeStamp();
_encodedImage._encodedHeight = inputImage.Height();
_encodedImage._encodedWidth = inputImage.Width();
if (inputImage.Length() > _encodedImage._size) {
// Allocating encoded memory.
if (_encodedImage._buffer != NULL) {
delete [] _encodedImage._buffer;
_encodedImage._buffer = NULL;
_encodedImage._size = 0;
}
const uint32_t newSize = CalcBufferSize(kI420,
_encodedImage._encodedWidth,
_encodedImage._encodedHeight);
uint8_t* newBuffer = new uint8_t[newSize];
if (newBuffer == NULL) {
return WEBRTC_VIDEO_CODEC_MEMORY;
}
_encodedImage._size = newSize;
_encodedImage._buffer = newBuffer;
}
memcpy(_encodedImage._buffer, inputImage.Buffer(), inputImage.Length());
_encodedImage._length = inputImage.Length();
_encodedCompleteCallback->Encoded(_encodedImage);
return WEBRTC_VIDEO_CODEC_OK;
}
int
I420Encoder::RegisterEncodeCompleteCallback(EncodedImageCallback* callback) {
_encodedCompleteCallback = callback;
return WEBRTC_VIDEO_CODEC_OK;
}
I420Decoder::I420Decoder():
_decodedImage(),
_width(0),
_height(0),
_inited(false),
_decodeCompleteCallback(NULL)
{}
I420Decoder::~I420Decoder() {
Release();
}
int
I420Decoder::Reset() {
return WEBRTC_VIDEO_CODEC_OK;
}
int
I420Decoder::InitDecode(const VideoCodec* codecSettings,
int /*numberOfCores */) {
if (codecSettings == NULL) {
return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
} else if (codecSettings->width < 1 || codecSettings->height < 1) {
return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
}
_width = codecSettings->width;
_height = codecSettings->height;
_inited = true;
return WEBRTC_VIDEO_CODEC_OK;
}
int
I420Decoder::Decode(const EncodedImage& inputImage,
bool /*missingFrames*/,
const RTPFragmentationHeader* /*fragmentation*/,
const CodecSpecificInfo* /*codecSpecificInfo*/,
int64_t /*renderTimeMs*/) {
if (inputImage._buffer == NULL) {
return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
}
if (_decodeCompleteCallback == NULL) {
return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
}
if (inputImage._length <= 0) {
return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
}
if (!_inited) {
return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
}
// Set decoded image parameters.
if (_decodedImage.CopyFrame(inputImage._length, inputImage._buffer) < 0) {
return WEBRTC_VIDEO_CODEC_MEMORY;
}
_decodedImage.SetHeight(_height);
_decodedImage.SetWidth(_width);
_decodedImage.SetTimeStamp(inputImage._timeStamp);
_decodeCompleteCallback->Decoded(_decodedImage);
return WEBRTC_VIDEO_CODEC_OK;
}
int
I420Decoder::RegisterDecodeCompleteCallback(DecodedImageCallback* callback) {
_decodeCompleteCallback = callback;
return WEBRTC_VIDEO_CODEC_OK;
}
int
I420Decoder::Release() {
_decodedImage.Free();
_inited = false;
return WEBRTC_VIDEO_CODEC_OK;
}
}

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# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'webrtc_i420',
'type': '<(library)',
'dependencies': [
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
],
'include_dirs': [
'../interface',
'../../../interface',
'../../../../../../common_video/interface',
],
'direct_dependent_settings': {
'include_dirs': [
'../interface',
'../../../../../../common_video/interface',
],
},
'sources': [
'../interface/i420.h',
'i420.cc',
],
},
],
}
# Local Variables:
# tab-width:2
# indent-tabs-mode:nil
# End:
# vim: set expandtab tabstop=2 shiftwidth=2:

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_MOCK_MOCK_VIDEO_CODEC_INTERFACE_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_MOCK_MOCK_VIDEO_CODEC_INTERFACE_H_
#include <string>
#include "gmock/gmock.h"
#include "modules/video_coding/codecs/interface/video_codec_interface.h"
#include "typedefs.h"
namespace webrtc {
class MockEncodedImageCallback : public EncodedImageCallback {
public:
MOCK_METHOD3(Encoded,
WebRtc_Word32(EncodedImage& encodedImage,
const CodecSpecificInfo* codecSpecificInfo,
const RTPFragmentationHeader* fragmentation));
};
class MockVideoEncoder : public VideoEncoder {
public:
MOCK_CONST_METHOD2(Version,
WebRtc_Word32(WebRtc_Word8 *version,
WebRtc_Word32 length));
MOCK_METHOD3(InitEncode,
WebRtc_Word32(const VideoCodec* codecSettings,
WebRtc_Word32 numberOfCores,
WebRtc_UWord32 maxPayloadSize));
MOCK_METHOD3(Encode,
WebRtc_Word32(const VideoFrame& inputImage,
const CodecSpecificInfo* codecSpecificInfo,
const std::vector<VideoFrameType>* frame_types));
MOCK_METHOD1(RegisterEncodeCompleteCallback,
WebRtc_Word32(EncodedImageCallback* callback));
MOCK_METHOD0(Release, WebRtc_Word32());
MOCK_METHOD0(Reset, WebRtc_Word32());
MOCK_METHOD2(SetChannelParameters, WebRtc_Word32(WebRtc_UWord32 packetLoss,
int rtt));
MOCK_METHOD2(SetRates,
WebRtc_Word32(WebRtc_UWord32 newBitRate,
WebRtc_UWord32 frameRate));
MOCK_METHOD1(SetPeriodicKeyFrames, WebRtc_Word32(bool enable));
MOCK_METHOD2(CodecConfigParameters,
WebRtc_Word32(WebRtc_UWord8* /*buffer*/, WebRtc_Word32));
};
class MockDecodedImageCallback : public DecodedImageCallback {
public:
MOCK_METHOD1(Decoded,
WebRtc_Word32(VideoFrame& decodedImage));
MOCK_METHOD1(ReceivedDecodedReferenceFrame,
WebRtc_Word32(const WebRtc_UWord64 pictureId));
MOCK_METHOD1(ReceivedDecodedFrame,
WebRtc_Word32(const WebRtc_UWord64 pictureId));
};
class MockVideoDecoder : public VideoDecoder {
public:
MOCK_METHOD2(InitDecode,
WebRtc_Word32(const VideoCodec* codecSettings,
WebRtc_Word32 numberOfCores));
MOCK_METHOD5(Decode,
WebRtc_Word32(const EncodedImage& inputImage,
bool missingFrames,
const RTPFragmentationHeader* fragmentation,
const CodecSpecificInfo* codecSpecificInfo,
WebRtc_Word64 renderTimeMs));
MOCK_METHOD1(RegisterDecodeCompleteCallback,
WebRtc_Word32(DecodedImageCallback* callback));
MOCK_METHOD0(Release, WebRtc_Word32());
MOCK_METHOD0(Reset, WebRtc_Word32());
MOCK_METHOD2(SetCodecConfigParameters,
WebRtc_Word32(const WebRtc_UWord8* /*buffer*/, WebRtc_Word32));
MOCK_METHOD0(Copy, VideoDecoder*());
};
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_MOCK_MOCK_VIDEO_CODEC_INTERFACE_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H
#include <vector>
#include "common_types.h"
#include "modules/interface/module_common_types.h"
#include "modules/video_coding/codecs/interface/video_error_codes.h"
#include "common_video/interface/video_image.h"
#include "typedefs.h"
namespace webrtc
{
class RTPFragmentationHeader; // forward declaration
// Note: if any pointers are added to this struct, it must be fitted
// with a copy-constructor. See below.
struct CodecSpecificInfoVP8
{
bool hasReceivedSLI;
WebRtc_UWord8 pictureIdSLI;
bool hasReceivedRPSI;
WebRtc_UWord64 pictureIdRPSI;
WebRtc_Word16 pictureId; // negative value to skip pictureId
bool nonReference;
WebRtc_UWord8 simulcastIdx;
WebRtc_UWord8 temporalIdx;
bool layerSync;
int tl0PicIdx; // Negative value to skip tl0PicIdx
WebRtc_Word8 keyIdx; // negative value to skip keyIdx
};
union CodecSpecificInfoUnion
{
CodecSpecificInfoVP8 VP8;
};
// Note: if any pointers are added to this struct or its sub-structs, it
// must be fitted with a copy-constructor. This is because it is copied
// in the copy-constructor of VCMEncodedFrame.
struct CodecSpecificInfo
{
VideoCodecType codecType;
CodecSpecificInfoUnion codecSpecific;
};
class EncodedImageCallback
{
public:
virtual ~EncodedImageCallback() {};
// Callback function which is called when an image has been encoded.
//
// Input:
// - encodedImage : The encoded image
//
// Return value : > 0, signals to the caller that one or more future frames
// should be dropped to keep bit rate or frame rate.
// = 0, if OK.
// < 0, on error.
virtual WebRtc_Word32
Encoded(EncodedImage& encodedImage,
const CodecSpecificInfo* codecSpecificInfo = NULL,
const RTPFragmentationHeader* fragmentation = NULL) = 0;
};
class VideoEncoder
{
public:
virtual ~VideoEncoder() {};
// Initialize the encoder with the information from the VideoCodec.
//
// Input:
// - codecSettings : Codec settings
// - numberOfCores : Number of cores available for the encoder
// - maxPayloadSize : The maximum size each payload is allowed
// to have. Usually MTU - overhead.
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
virtual WebRtc_Word32 InitEncode(const VideoCodec* codecSettings, WebRtc_Word32 numberOfCores, WebRtc_UWord32 maxPayloadSize) = 0;
// Encode an I420 image (as a part of a video stream). The encoded image
// will be returned to the user through the encode complete callback.
//
// Input:
// - inputImage : Image to be encoded
// - codecSpecificInfo : Pointer to codec specific data
// - frame_types : The frame type to encode
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0
// otherwise.
virtual WebRtc_Word32 Encode(
const VideoFrame& inputImage,
const CodecSpecificInfo* codecSpecificInfo,
const std::vector<VideoFrameType>* frame_types) = 0;
// Register an encode complete callback object.
//
// Input:
// - callback : Callback object which handles encoded images.
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
virtual WebRtc_Word32 RegisterEncodeCompleteCallback(EncodedImageCallback* callback) = 0;
// Free encoder memory.
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
virtual WebRtc_Word32 Release() = 0;
// Inform the encoder about the packet loss and round trip time on the
// network used to decide the best pattern and signaling.
//
// - packetLoss : Fraction lost (loss rate in percent =
// 100 * packetLoss / 255)
// - rtt : Round-trip time in milliseconds
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
virtual WebRtc_Word32 SetChannelParameters(WebRtc_UWord32 packetLoss,
int rtt) = 0;
// Inform the encoder about the new target bit rate.
//
// - newBitRate : New target bit rate
// - frameRate : The target frame rate
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
virtual WebRtc_Word32 SetRates(WebRtc_UWord32 newBitRate, WebRtc_UWord32 frameRate) = 0;
// Use this function to enable or disable periodic key frames. Can be useful for codecs
// which have other ways of stopping error propagation.
//
// - enable : Enable or disable periodic key frames
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
virtual WebRtc_Word32 SetPeriodicKeyFrames(bool enable) { return WEBRTC_VIDEO_CODEC_ERROR; }
// Codec configuration data to send out-of-band, i.e. in SIP call setup
//
// - buffer : Buffer pointer to where the configuration data
// should be stored
// - size : The size of the buffer in bytes
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
virtual WebRtc_Word32 CodecConfigParameters(WebRtc_UWord8* /*buffer*/, WebRtc_Word32 /*size*/) { return WEBRTC_VIDEO_CODEC_ERROR; }
};
class DecodedImageCallback
{
public:
virtual ~DecodedImageCallback() {};
// Callback function which is called when an image has been decoded.
//
// Input:
// - decodedImage : The decoded image.
//
// Return value : 0 if OK, < 0 otherwise.
virtual WebRtc_Word32 Decoded(VideoFrame& decodedImage) = 0;
virtual WebRtc_Word32 ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId) {return -1;}
virtual WebRtc_Word32 ReceivedDecodedFrame(const WebRtc_UWord64 pictureId) {return -1;}
};
class VideoDecoder
{
public:
virtual ~VideoDecoder() {};
// Initialize the decoder with the information from the VideoCodec.
//
// Input:
// - inst : Codec settings
// - numberOfCores : Number of cores available for the decoder
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
virtual WebRtc_Word32 InitDecode(const VideoCodec* codecSettings, WebRtc_Word32 numberOfCores) = 0;
// Decode encoded image (as a part of a video stream). The decoded image
// will be returned to the user through the decode complete callback.
//
// Input:
// - inputImage : Encoded image to be decoded
// - missingFrames : True if one or more frames have been lost
// since the previous decode call.
// - fragmentation : Specifies where the encoded frame can be
// split into separate fragments. The meaning
// of fragment is codec specific, but often
// means that each fragment is decodable by
// itself.
// - codecSpecificInfo : Pointer to codec specific data
// - renderTimeMs : System time to render in milliseconds. Only
// used by decoders with internal rendering.
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
virtual WebRtc_Word32
Decode(const EncodedImage& inputImage,
bool missingFrames,
const RTPFragmentationHeader* fragmentation,
const CodecSpecificInfo* codecSpecificInfo = NULL,
WebRtc_Word64 renderTimeMs = -1) = 0;
// Register an decode complete callback object.
//
// Input:
// - callback : Callback object which handles decoded images.
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
virtual WebRtc_Word32 RegisterDecodeCompleteCallback(DecodedImageCallback* callback) = 0;
// Free decoder memory.
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
virtual WebRtc_Word32 Release() = 0;
// Reset decoder state and prepare for a new call.
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
virtual WebRtc_Word32 Reset() = 0;
// Codec configuration data sent out-of-band, i.e. in SIP call setup
//
// Input/Output:
// - buffer : Buffer pointer to the configuration data
// - size : The size of the configuration data in
// bytes
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
virtual WebRtc_Word32 SetCodecConfigParameters(const WebRtc_UWord8* /*buffer*/, WebRtc_Word32 /*size*/) { return WEBRTC_VIDEO_CODEC_ERROR; }
// Create a copy of the codec and its internal state.
//
// Return value : A copy of the instance if OK, NULL otherwise.
virtual VideoDecoder* Copy() { return NULL; }
};
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_ERROR_CODES_H
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_ERROR_CODES_H
// NOTE: in sync with video_coding_module_defines.h
// Define return values
#define WEBRTC_VIDEO_CODEC_REQUEST_SLI 2
#define WEBRTC_VIDEO_CODEC_NO_OUTPUT 1
#define WEBRTC_VIDEO_CODEC_OK 0
#define WEBRTC_VIDEO_CODEC_ERROR -1
#define WEBRTC_VIDEO_CODEC_LEVEL_EXCEEDED -2
#define WEBRTC_VIDEO_CODEC_MEMORY -3
#define WEBRTC_VIDEO_CODEC_ERR_PARAMETER -4
#define WEBRTC_VIDEO_CODEC_ERR_SIZE -5
#define WEBRTC_VIDEO_CODEC_TIMEOUT -6
#define WEBRTC_VIDEO_CODEC_UNINITIALIZED -7
#define WEBRTC_VIDEO_CODEC_ERR_REQUEST_SLI -12
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_ERROR_CODES_H

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_MOCK_MOCK_PACKET_MANIPULATOR_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_MOCK_MOCK_PACKET_MANIPULATOR_H_
#include "modules/video_coding/codecs/test/packet_manipulator.h"
#include <string>
#include "common_video/interface/video_image.h"
#include "gmock/gmock.h"
#include "typedefs.h"
namespace webrtc {
namespace test {
class MockPacketManipulator : public PacketManipulator {
public:
MOCK_METHOD1(ManipulatePackets, int(webrtc::EncodedImage* encoded_image));
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_MOCK_MOCK_PACKET_MANIPULATOR_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/video_coding/codecs/test/packet_manipulator.h"
#include <cassert>
#include <cstdio>
namespace webrtc {
namespace test {
PacketManipulatorImpl::PacketManipulatorImpl(PacketReader* packet_reader,
const NetworkingConfig& config,
bool verbose)
: packet_reader_(packet_reader),
config_(config),
active_burst_packets_(0),
critsect_(CriticalSectionWrapper::CreateCriticalSection()),
random_seed_(1),
verbose_(verbose) {
assert(packet_reader);
}
PacketManipulatorImpl::~PacketManipulatorImpl() {
delete critsect_;
}
int PacketManipulatorImpl::ManipulatePackets(
webrtc::EncodedImage* encoded_image) {
assert(encoded_image);
int nbr_packets_dropped = 0;
// There's no need to build a copy of the image data since viewing an
// EncodedImage object, setting the length to a new lower value represents
// that everything is dropped after that position in the byte array.
// EncodedImage._size is the allocated bytes.
// EncodedImage._length is how many that are filled with data.
int new_length = 0;
packet_reader_->InitializeReading(encoded_image->_buffer,
encoded_image->_length,
config_.packet_size_in_bytes);
WebRtc_UWord8* packet = NULL;
int nbr_bytes_to_read;
// keep track of if we've lost any packets, since then we shall loose
// the remains of the current frame:
bool packet_loss_has_occurred = false;
while ((nbr_bytes_to_read = packet_reader_->NextPacket(&packet)) > 0) {
// Check if we're currently in a packet loss burst that is not completed:
if (active_burst_packets_ > 0) {
active_burst_packets_--;
nbr_packets_dropped++;
} else if (RandomUniform() < config_.packet_loss_probability ||
packet_loss_has_occurred) {
packet_loss_has_occurred = true;
nbr_packets_dropped++;
if (config_.packet_loss_mode == kBurst) {
// Initiate a new burst
active_burst_packets_ = config_.packet_loss_burst_length - 1;
}
} else {
new_length += nbr_bytes_to_read;
}
}
encoded_image->_length = new_length;
if (nbr_packets_dropped > 0) {
// Must set completeFrame to false to inform the decoder about this:
encoded_image->_completeFrame = false;
if (verbose_) {
printf("Dropped %d packets for frame %d (frame length: %d)\n",
nbr_packets_dropped, encoded_image->_timeStamp,
encoded_image->_length);
}
}
return nbr_packets_dropped;
}
void PacketManipulatorImpl::InitializeRandomSeed(unsigned int seed) {
random_seed_ = seed;
}
inline double PacketManipulatorImpl::RandomUniform() {
// Use the previous result as new seed before each rand() call. Doing this
// it doesn't matter if other threads are calling rand() since we'll always
// get the same behavior as long as we're using a fixed initial seed.
critsect_->Enter();
srand(random_seed_);
random_seed_ = std::rand();
critsect_->Leave();
return (random_seed_ + 1.0)/(RAND_MAX + 1.0);
}
const char* PacketLossModeToStr(PacketLossMode e) {
switch (e) {
case kUniform:
return "Uniform";
case kBurst:
return "Burst";
default:
assert(false);
return "Unknown";
}
}
} // namespace test
} // namespace webrtcc

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_PACKET_MANIPULATOR_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_PACKET_MANIPULATOR_H_
#include <cstdlib>
#include "modules/video_coding/codecs/interface/video_codec_interface.h"
#include "system_wrappers/interface/critical_section_wrapper.h"
#include "testsupport/packet_reader.h"
namespace webrtc {
namespace test {
// Which mode the packet loss shall be performed according to.
enum PacketLossMode {
// Drops packets with a configured probability independently for each packet
kUniform,
// Drops packets similar to uniform but when a packet is being dropped,
// the number of lost packets in a row is equal to the configured burst
// length.
kBurst
};
// Returns a string representation of the enum value.
const char* PacketLossModeToStr(PacketLossMode e);
// Contains configurations related to networking and simulation of
// scenarios caused by network interference.
struct NetworkingConfig {
NetworkingConfig()
: packet_size_in_bytes(1500), max_payload_size_in_bytes(1440),
packet_loss_mode(kUniform), packet_loss_probability(0.0),
packet_loss_burst_length(1) {
}
// Packet size in bytes. Default: 1500 bytes.
int packet_size_in_bytes;
// Encoder specific setting of maximum size in bytes of each payload.
// Default: 1440 bytes.
int max_payload_size_in_bytes;
// Packet loss mode. Two different packet loss models are supported:
// uniform or burst. This setting has no effect unless
// packet_loss_probability is >0.
// Default: uniform.
PacketLossMode packet_loss_mode;
// Packet loss probability. A value between 0.0 and 1.0 that defines the
// probability of a packet being lost. 0.1 means 10% and so on.
// Default: 0 (no loss).
double packet_loss_probability;
// Packet loss burst length. Defines how many packets will be lost in a burst
// when a packet has been decided to be lost. Must be >=1. Default: 1.
int packet_loss_burst_length;
};
// Class for simulating packet loss on the encoded frame data.
// When a packet loss has occurred in a frame, the remaining data in that
// frame is lost (even if burst length is only a single packet).
// TODO(kjellander): Support discarding only individual packets in the frame
// when CL 172001 has been submitted. This also requires a correct
// fragmentation header to be passed to the decoder.
//
// To get a repeatable packet drop pattern, re-initialize the random seed
// using InitializeRandomSeed before each test run.
class PacketManipulator {
public:
virtual ~PacketManipulator() {}
// Manipulates the data of the encoded_image to simulate parts being lost
// during transport.
// If packets are dropped from frame data, the completedFrame field will be
// set to false.
// Returns the number of packets being dropped.
virtual int
ManipulatePackets(webrtc::EncodedImage* encoded_image) = 0;
};
class PacketManipulatorImpl : public PacketManipulator {
public:
PacketManipulatorImpl(PacketReader* packet_reader,
const NetworkingConfig& config,
bool verbose);
virtual ~PacketManipulatorImpl();
virtual int ManipulatePackets(webrtc::EncodedImage* encoded_image);
virtual void InitializeRandomSeed(unsigned int seed);
protected:
// Returns a uniformly distributed random value between 0.0 and 1.0
virtual double RandomUniform();
private:
PacketReader* packet_reader_;
const NetworkingConfig& config_;
// Used to simulate a burst over several frames.
int active_burst_packets_;
CriticalSectionWrapper* critsect_;
unsigned int random_seed_;
bool verbose_;
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_PACKET_MANIPULATOR_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/video_coding/codecs/test/packet_manipulator.h"
#include <queue>
#include "gtest/gtest.h"
#include "modules/video_coding/codecs/interface/video_codec_interface.h"
#include "modules/video_coding/codecs/test/predictive_packet_manipulator.h"
#include "testsupport/unittest_utils.h"
#include "typedefs.h"
namespace webrtc {
namespace test {
const double kNeverDropProbability = 0.0;
const double kAlwaysDropProbability = 1.0;
const int kBurstLength = 1;
class PacketManipulatorTest: public PacketRelatedTest {
protected:
PacketReader packet_reader_;
EncodedImage image_;
NetworkingConfig drop_config_;
NetworkingConfig no_drop_config_;
PacketManipulatorTest() {
image_._buffer = packet_data_;
image_._length = kPacketDataLength;
image_._size = kPacketDataLength;
drop_config_.packet_size_in_bytes = kPacketSizeInBytes;
drop_config_.packet_loss_probability = kAlwaysDropProbability;
drop_config_.packet_loss_burst_length = kBurstLength;
drop_config_.packet_loss_mode = kUniform;
no_drop_config_.packet_size_in_bytes = kPacketSizeInBytes;
no_drop_config_.packet_loss_probability = kNeverDropProbability;
no_drop_config_.packet_loss_burst_length = kBurstLength;
no_drop_config_.packet_loss_mode = kUniform;
}
virtual ~PacketManipulatorTest() {}
void SetUp() {
PacketRelatedTest::SetUp();
}
void TearDown() {
PacketRelatedTest::TearDown();
}
void VerifyPacketLoss(int expected_nbr_packets_dropped,
int actual_nbr_packets_dropped,
int expected_packet_data_length,
WebRtc_UWord8* expected_packet_data,
EncodedImage& actual_image) {
EXPECT_EQ(expected_nbr_packets_dropped, actual_nbr_packets_dropped);
EXPECT_EQ(expected_packet_data_length, static_cast<int>(image_._length));
EXPECT_EQ(0, memcmp(expected_packet_data, actual_image._buffer,
expected_packet_data_length));
}
};
TEST_F(PacketManipulatorTest, Constructor) {
PacketManipulatorImpl manipulator(&packet_reader_, no_drop_config_, false);
}
TEST_F(PacketManipulatorTest, DropNone) {
PacketManipulatorImpl manipulator(&packet_reader_, no_drop_config_, false);
int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
VerifyPacketLoss(0, nbr_packets_dropped, kPacketDataLength,
packet_data_, image_);
}
TEST_F(PacketManipulatorTest, UniformDropNoneSmallFrame) {
int data_length = 400; // smaller than the packet size
image_._length = data_length;
PacketManipulatorImpl manipulator(&packet_reader_, no_drop_config_, false);
int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
VerifyPacketLoss(0, nbr_packets_dropped, data_length,
packet_data_, image_);
}
TEST_F(PacketManipulatorTest, UniformDropAll) {
PacketManipulatorImpl manipulator(&packet_reader_, drop_config_, false);
int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
VerifyPacketLoss(kPacketDataNumberOfPackets, nbr_packets_dropped,
0, packet_data_, image_);
}
// Use our customized test class to make the second packet being lost
TEST_F(PacketManipulatorTest, UniformDropSinglePacket) {
drop_config_.packet_loss_probability = 0.5;
PredictivePacketManipulator manipulator(&packet_reader_, drop_config_);
manipulator.AddRandomResult(1.0);
manipulator.AddRandomResult(0.3); // less than 0.5 will cause packet loss
manipulator.AddRandomResult(1.0);
// Execute the test target method:
int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
// Since we setup the predictive packet manipulator, it will throw away the
// second packet. The third packet is also lost because when we have lost one,
// the remains shall also be discarded (in the current implementation).
VerifyPacketLoss(2, nbr_packets_dropped, kPacketSizeInBytes, packet1_,
image_);
}
// Use our customized test class to make the second packet being lost
TEST_F(PacketManipulatorTest, BurstDropNinePackets) {
// Create a longer packet data structure (10 packets)
const int kNbrPackets = 10;
const int kDataLength = kPacketSizeInBytes * kNbrPackets;
WebRtc_UWord8 data[kDataLength];
WebRtc_UWord8* data_pointer = data;
// Fill with 0s, 1s and so on to be able to easily verify which were dropped:
for (int i = 0; i < kNbrPackets; ++i) {
memset(data_pointer + i * kPacketSizeInBytes, i, kPacketSizeInBytes);
}
// Overwrite the defaults from the test fixture:
image_._buffer = data;
image_._length = kDataLength;
image_._size = kDataLength;
drop_config_.packet_loss_probability = 0.5;
drop_config_.packet_loss_burst_length = 5;
drop_config_.packet_loss_mode = kBurst;
PredictivePacketManipulator manipulator(&packet_reader_, drop_config_);
manipulator.AddRandomResult(1.0);
manipulator.AddRandomResult(0.3); // less than 0.5 will cause packet loss
for (int i = 0; i < kNbrPackets - 2; ++i) {
manipulator.AddRandomResult(1.0);
}
// Execute the test target method:
int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
// Should discard every packet after the first one.
VerifyPacketLoss(9, nbr_packets_dropped, kPacketSizeInBytes, data, image_);
}
} // namespace test
} // namespace webrtc

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/video_coding/codecs/test/predictive_packet_manipulator.h"
#include <cassert>
#include <cstdio>
#include "testsupport/packet_reader.h"
namespace webrtc {
namespace test {
PredictivePacketManipulator::PredictivePacketManipulator(
PacketReader* packet_reader, const NetworkingConfig& config)
: PacketManipulatorImpl(packet_reader, config, false) {
}
PredictivePacketManipulator::~PredictivePacketManipulator() {
}
void PredictivePacketManipulator::AddRandomResult(double result) {
assert(result >= 0.0 && result <= 1.0);
random_results_.push(result);
}
double PredictivePacketManipulator::RandomUniform() {
if(random_results_.size() == 0u) {
fprintf(stderr, "No more stored results, please make sure AddRandomResult()"
"is called same amount of times you're going to invoke the "
"RandomUniform() function, i.e. once per packet.\n");
assert(false);
}
double result = random_results_.front();
random_results_.pop();
return result;
}
} // namespace test
} // namespace webrtcc

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_PREDICTIVE_PACKET_MANIPULATOR_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_PREDICTIVE_PACKET_MANIPULATOR_H_
#include <queue>
#include "modules/video_coding/codecs/test/packet_manipulator.h"
#include "testsupport/packet_reader.h"
namespace webrtc {
namespace test {
// Predictive packet manipulator that allows for setup of the result of
// the random invocations.
class PredictivePacketManipulator : public PacketManipulatorImpl {
public:
PredictivePacketManipulator(PacketReader* packet_reader,
const NetworkingConfig& config);
virtual ~PredictivePacketManipulator();
// Adds a result. You must add at least the same number of results as the
// expected calls to the RandomUniform method. The results are added to a
// FIFO queue so they will be returned in the same order they were added.
// Result parameter must be 0.0 to 1.0.
void AddRandomResult(double result);
protected:
// Returns a uniformly distributed random value between 0.0 and 1.0
virtual double RandomUniform();
private:
std::queue<double> random_results_;
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_PREDICTIVE_PACKET_MANIPULATOR_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/video_coding/codecs/test/stats.h"
#include <algorithm> // min_element, max_element
#include <cassert>
#include <cstdio>
namespace webrtc {
namespace test {
Stats::Stats() {}
Stats::~Stats() {}
bool LessForEncodeTime(const FrameStatistic& s1, const FrameStatistic& s2) {
return s1.encode_time_in_us < s2.encode_time_in_us;
}
bool LessForDecodeTime(const FrameStatistic& s1, const FrameStatistic& s2) {
return s1.decode_time_in_us < s2.decode_time_in_us;
}
bool LessForEncodedSize(const FrameStatistic& s1, const FrameStatistic& s2) {
return s1.encoded_frame_length_in_bytes < s2.encoded_frame_length_in_bytes;
}
bool LessForBitRate(const FrameStatistic& s1, const FrameStatistic& s2) {
return s1.bit_rate_in_kbps < s2.bit_rate_in_kbps;
}
FrameStatistic& Stats::NewFrame(int frame_number) {
assert(frame_number >= 0);
FrameStatistic stat;
stat.frame_number = frame_number;
stats_.push_back(stat);
return stats_[frame_number];
}
void Stats::PrintSummary() {
printf("Processing summary:\n");
if (stats_.size() == 0) {
printf("No frame statistics have been logged yet.\n");
return;
}
// Calculate min, max, average and total encoding time
int total_encoding_time_in_us = 0;
int total_decoding_time_in_us = 0;
int total_encoded_frames_lengths = 0;
int total_encoded_key_frames_lengths = 0;
int total_encoded_nonkey_frames_lengths = 0;
int nbr_keyframes = 0;
int nbr_nonkeyframes = 0;
for (FrameStatisticsIterator it = stats_.begin();
it != stats_.end(); ++it) {
total_encoding_time_in_us += it->encode_time_in_us;
total_decoding_time_in_us += it->decode_time_in_us;
total_encoded_frames_lengths += it->encoded_frame_length_in_bytes;
if (it->frame_type == webrtc::kKeyFrame) {
total_encoded_key_frames_lengths += it->encoded_frame_length_in_bytes;
nbr_keyframes++;
} else {
total_encoded_nonkey_frames_lengths += it->encoded_frame_length_in_bytes;
nbr_nonkeyframes++;
}
}
FrameStatisticsIterator frame;
// ENCODING
printf("Encoding time:\n");
frame = std::min_element(stats_.begin(),
stats_.end(), LessForEncodeTime);
printf(" Min : %7d us (frame %d)\n",
frame->encode_time_in_us, frame->frame_number);
frame = std::max_element(stats_.begin(),
stats_.end(), LessForEncodeTime);
printf(" Max : %7d us (frame %d)\n",
frame->encode_time_in_us, frame->frame_number);
printf(" Average : %7d us\n",
static_cast<int>(total_encoding_time_in_us / stats_.size()));
// DECODING
printf("Decoding time:\n");
// only consider frames that were successfully decoded (packet loss may cause
// failures)
std::vector<FrameStatistic> decoded_frames;
for (std::vector<FrameStatistic>::iterator it = stats_.begin();
it != stats_.end(); ++it) {
if (it->decoding_successful) {
decoded_frames.push_back(*it);
}
}
if (decoded_frames.size() == 0) {
printf("No successfully decoded frames exist in this statistics.\n");
} else {
frame = std::min_element(decoded_frames.begin(),
decoded_frames.end(), LessForDecodeTime);
printf(" Min : %7d us (frame %d)\n",
frame->decode_time_in_us, frame->frame_number);
frame = std::max_element(decoded_frames.begin(),
decoded_frames.end(), LessForDecodeTime);
printf(" Max : %7d us (frame %d)\n",
frame->decode_time_in_us, frame->frame_number);
printf(" Average : %7d us\n",
static_cast<int>(total_decoding_time_in_us / decoded_frames.size()));
printf(" Failures: %d frames failed to decode.\n",
static_cast<int>(stats_.size() - decoded_frames.size()));
}
// SIZE
printf("Frame sizes:\n");
frame = std::min_element(stats_.begin(),
stats_.end(), LessForEncodedSize);
printf(" Min : %7d bytes (frame %d)\n",
frame->encoded_frame_length_in_bytes, frame->frame_number);
frame = std::max_element(stats_.begin(),
stats_.end(), LessForEncodedSize);
printf(" Max : %7d bytes (frame %d)\n",
frame->encoded_frame_length_in_bytes, frame->frame_number);
printf(" Average : %7d bytes\n",
static_cast<int>(total_encoded_frames_lengths / stats_.size()));
if (nbr_keyframes > 0) {
printf(" Average key frame size : %7d bytes (%d keyframes)\n",
total_encoded_key_frames_lengths / nbr_keyframes,
nbr_keyframes);
}
if (nbr_nonkeyframes > 0) {
printf(" Average non-key frame size: %7d bytes (%d frames)\n",
total_encoded_nonkey_frames_lengths / nbr_nonkeyframes,
nbr_nonkeyframes);
}
// BIT RATE
printf("Bit rates:\n");
frame = std::min_element(stats_.begin(),
stats_.end(), LessForBitRate);
printf(" Min bit rate: %7d kbps (frame %d)\n",
frame->bit_rate_in_kbps, frame->frame_number);
frame = std::max_element(stats_.begin(),
stats_.end(), LessForBitRate);
printf(" Max bit rate: %7d kbps (frame %d)\n",
frame->bit_rate_in_kbps, frame->frame_number);
printf("\n");
printf("Total encoding time : %7d ms.\n",
total_encoding_time_in_us / 1000);
printf("Total decoding time : %7d ms.\n",
total_decoding_time_in_us / 1000);
printf("Total processing time: %7d ms.\n",
(total_encoding_time_in_us + total_decoding_time_in_us) / 1000);
}
} // namespace test
} // namespace webrtc

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_STATS_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_STATS_H_
#include <vector>
#include "common_video/interface/video_image.h"
namespace webrtc {
namespace test {
// Contains statistics of a single frame that has been processed.
struct FrameStatistic {
FrameStatistic() :
encoding_successful(false), decoding_successful(false),
encode_return_code(0), decode_return_code(0),
encode_time_in_us(0), decode_time_in_us(0),
frame_number(0), packets_dropped(0), total_packets(0),
bit_rate_in_kbps(0), encoded_frame_length_in_bytes(0),
frame_type(kDeltaFrame) {
};
bool encoding_successful;
bool decoding_successful;
int encode_return_code;
int decode_return_code;
int encode_time_in_us;
int decode_time_in_us;
int frame_number;
// How many packets were discarded of the encoded frame data (if any)
int packets_dropped;
int total_packets;
// Current bit rate. Calculated out of the size divided with the time
// interval per frame.
int bit_rate_in_kbps;
// Copied from EncodedImage
int encoded_frame_length_in_bytes;
webrtc::VideoFrameType frame_type;
};
// Handles statistics from a single video processing run.
// Contains calculation methods for interesting metrics from these stats.
class Stats {
public:
typedef std::vector<FrameStatistic>::iterator FrameStatisticsIterator;
Stats();
virtual ~Stats();
// Add a new statistic data object.
// The frame number must be incrementing and start at zero in order to use
// it as an index for the frame_statistics_ vector.
// Returns the newly created statistic object.
FrameStatistic& NewFrame(int frame_number);
// Prints a summary of all the statistics that have been gathered during the
// processing
void PrintSummary();
std::vector<FrameStatistic> stats_;
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_STATS_H_

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/video_coding/codecs/test/stats.h"
#include "gtest/gtest.h"
#include "typedefs.h"
namespace webrtc {
namespace test {
class StatsTest: public testing::Test {
protected:
StatsTest() {
}
virtual ~StatsTest() {
}
void SetUp() {
stats_ = new Stats();
}
void TearDown() {
delete stats_;
}
Stats* stats_;
};
// Test empty object
TEST_F(StatsTest, Uninitialized) {
EXPECT_EQ(0u, stats_->stats_.size());
stats_->PrintSummary(); // should not crash
}
// Add single frame stats and verify
TEST_F(StatsTest, AddOne) {
stats_->NewFrame(0u);
FrameStatistic* frameStat = &stats_->stats_[0];
EXPECT_EQ(0, frameStat->frame_number);
}
// Add multiple frame stats and verify
TEST_F(StatsTest, AddMany) {
int nbr_of_frames = 1000;
for (int i = 0; i < nbr_of_frames; ++i) {
FrameStatistic& frameStat = stats_->NewFrame(i);
EXPECT_EQ(i, frameStat.frame_number);
}
EXPECT_EQ(nbr_of_frames, static_cast<int>(stats_->stats_.size()));
stats_->PrintSummary(); // should not crash
}
} // namespace test
} // namespace webrtc

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# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'conditions': [
['include_tests==1', {
'targets': [
{
'target_name': 'video_codecs_test_framework',
'type': '<(library)',
'dependencies': [
'<(webrtc_root)/test/test.gyp:test_support',
],
'sources': [
'mock/mock_packet_manipulator.h',
'packet_manipulator.h',
'packet_manipulator.cc',
'predictive_packet_manipulator.h',
'predictive_packet_manipulator.cc',
'stats.h',
'stats.cc',
'videoprocessor.h',
'videoprocessor.cc',
],
},
{
'target_name': 'video_codecs_test_framework_unittests',
'type': 'executable',
'dependencies': [
'video_codecs_test_framework',
'webrtc_video_coding',
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/test/test.gyp:test_support_main',
],
'sources': [
'packet_manipulator_unittest.cc',
'stats_unittest.cc',
'videoprocessor_unittest.cc',
],
},
{
'target_name': 'video_codecs_test_framework_integrationtests',
'type': 'executable',
'dependencies': [
'video_codecs_test_framework',
'webrtc_video_coding',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/test/metrics.gyp:metrics',
'<(webrtc_root)/test/test.gyp:test_support_main',
'<(webrtc_vp8_dir)/vp8.gyp:webrtc_vp8',
],
'sources': [
'videoprocessor_integrationtest.cc',
],
},
], # targets
}], # include_tests
], # conditions
}

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/video_coding/codecs/test/videoprocessor.h"
#include <cassert>
#include <cstring>
#include <limits>
#include "system_wrappers/interface/cpu_info.h"
namespace webrtc {
namespace test {
VideoProcessorImpl::VideoProcessorImpl(webrtc::VideoEncoder* encoder,
webrtc::VideoDecoder* decoder,
FrameReader* frame_reader,
FrameWriter* frame_writer,
PacketManipulator* packet_manipulator,
const TestConfig& config,
Stats* stats)
: encoder_(encoder),
decoder_(decoder),
frame_reader_(frame_reader),
frame_writer_(frame_writer),
packet_manipulator_(packet_manipulator),
config_(config),
stats_(stats),
encode_callback_(NULL),
decode_callback_(NULL),
source_buffer_(NULL),
first_key_frame_has_been_excluded_(false),
last_frame_missing_(false),
initialized_(false),
encoded_frame_size_(0),
prev_time_stamp_(0),
num_dropped_frames_(0),
num_spatial_resizes_(0),
last_encoder_frame_width_(0),
last_encoder_frame_height_(0),
scaler_() {
assert(encoder);
assert(decoder);
assert(frame_reader);
assert(frame_writer);
assert(packet_manipulator);
assert(stats);
}
bool VideoProcessorImpl::Init() {
// Calculate a factor used for bit rate calculations:
bit_rate_factor_ = config_.codec_settings->maxFramerate * 0.001 * 8; // bits
int frame_length_in_bytes = frame_reader_->FrameLength();
// Initialize data structures used by the encoder/decoder APIs
source_buffer_ = new WebRtc_UWord8[frame_length_in_bytes];
last_successful_frame_buffer_ = new WebRtc_UWord8[frame_length_in_bytes];
// Set fixed properties common for all frames:
source_frame_.SetWidth(config_.codec_settings->width);
source_frame_.SetHeight(config_.codec_settings->height);
source_frame_.VerifyAndAllocate(frame_length_in_bytes);
source_frame_.SetLength(frame_length_in_bytes);
// To keep track of spatial resize actions by encoder.
last_encoder_frame_width_ = config_.codec_settings->width;
last_encoder_frame_height_ = config_.codec_settings->height;
// Setup required callbacks for the encoder/decoder:
encode_callback_ = new VideoProcessorEncodeCompleteCallback(this);
decode_callback_ = new VideoProcessorDecodeCompleteCallback(this);
WebRtc_Word32 register_result =
encoder_->RegisterEncodeCompleteCallback(encode_callback_);
if (register_result != WEBRTC_VIDEO_CODEC_OK) {
fprintf(stderr, "Failed to register encode complete callback, return code: "
"%d\n", register_result);
return false;
}
register_result = decoder_->RegisterDecodeCompleteCallback(decode_callback_);
if (register_result != WEBRTC_VIDEO_CODEC_OK) {
fprintf(stderr, "Failed to register decode complete callback, return code: "
"%d\n", register_result);
return false;
}
// Init the encoder and decoder
WebRtc_UWord32 nbr_of_cores = 1;
if (!config_.use_single_core) {
nbr_of_cores = CpuInfo::DetectNumberOfCores();
}
WebRtc_Word32 init_result =
encoder_->InitEncode(config_.codec_settings, nbr_of_cores,
config_.networking_config.max_payload_size_in_bytes);
if (init_result != WEBRTC_VIDEO_CODEC_OK) {
fprintf(stderr, "Failed to initialize VideoEncoder, return code: %d\n",
init_result);
return false;
}
init_result = decoder_->InitDecode(config_.codec_settings, nbr_of_cores);
if (init_result != WEBRTC_VIDEO_CODEC_OK) {
fprintf(stderr, "Failed to initialize VideoDecoder, return code: %d\n",
init_result);
return false;
}
if (config_.verbose) {
printf("Video Processor:\n");
printf(" #CPU cores used : %d\n", nbr_of_cores);
printf(" Total # of frames: %d\n", frame_reader_->NumberOfFrames());
printf(" Codec settings:\n");
printf(" Start bitrate : %d kbps\n",
config_.codec_settings->startBitrate);
printf(" Width : %d\n", config_.codec_settings->width);
printf(" Height : %d\n", config_.codec_settings->height);
}
initialized_ = true;
return true;
}
VideoProcessorImpl::~VideoProcessorImpl() {
delete[] source_buffer_;
delete[] last_successful_frame_buffer_;
encoder_->RegisterEncodeCompleteCallback(NULL);
delete encode_callback_;
decoder_->RegisterDecodeCompleteCallback(NULL);
delete decode_callback_;
}
void VideoProcessorImpl::SetRates(int bit_rate, int frame_rate) {
int set_rates_result = encoder_->SetRates(bit_rate, frame_rate);
assert(set_rates_result >= 0);
if (set_rates_result < 0) {
fprintf(stderr, "Failed to update encoder with new rate %d, "
"return code: %d\n", bit_rate, set_rates_result);
}
num_dropped_frames_ = 0;
num_spatial_resizes_ = 0;
}
int VideoProcessorImpl::EncodedFrameSize() {
return encoded_frame_size_;
}
int VideoProcessorImpl::NumberDroppedFrames() {
return num_dropped_frames_;
}
int VideoProcessorImpl::NumberSpatialResizes() {
return num_spatial_resizes_;
}
bool VideoProcessorImpl::ProcessFrame(int frame_number) {
assert(frame_number >=0);
if (!initialized_) {
fprintf(stderr, "Attempting to use uninitialized VideoProcessor!\n");
return false;
}
// |prev_time_stamp_| is used for getting number of dropped frames.
if (frame_number == 0) {
prev_time_stamp_ = -1;
}
if (frame_reader_->ReadFrame(source_buffer_)) {
// Copy the source frame to the newly read frame data.
// Length is common for all frames.
source_frame_.CopyFrame(source_frame_.Length(), source_buffer_);
// Ensure we have a new statistics data object we can fill:
FrameStatistic& stat = stats_->NewFrame(frame_number);
encode_start_ = TickTime::Now();
// Use the frame number as "timestamp" to identify frames
source_frame_.SetTimeStamp(frame_number);
// Decide if we're going to force a keyframe:
std::vector<VideoFrameType> frame_types(1, kDeltaFrame);
if (config_.keyframe_interval > 0 &&
frame_number % config_.keyframe_interval == 0) {
frame_types[0] = kKeyFrame;
}
// For dropped frames, we regard them as zero size encoded frames.
encoded_frame_size_ = 0;
WebRtc_Word32 encode_result = encoder_->Encode(source_frame_, NULL,
&frame_types);
if (encode_result != WEBRTC_VIDEO_CODEC_OK) {
fprintf(stderr, "Failed to encode frame %d, return code: %d\n",
frame_number, encode_result);
}
stat.encode_return_code = encode_result;
return true;
} else {
return false; // we've reached the last frame
}
}
void VideoProcessorImpl::FrameEncoded(EncodedImage* encoded_image) {
// Timestamp is frame number, so this gives us #dropped frames.
int num_dropped_from_prev_encode = encoded_image->_timeStamp -
prev_time_stamp_ - 1;
num_dropped_frames_ += num_dropped_from_prev_encode;
prev_time_stamp_ = encoded_image->_timeStamp;
if (num_dropped_from_prev_encode > 0) {
// For dropped frames, we write out the last decoded frame to avoid getting
// out of sync for the computation of PSNR and SSIM.
for (int i = 0; i < num_dropped_from_prev_encode; i++) {
frame_writer_->WriteFrame(last_successful_frame_buffer_);
}
}
// Frame is not dropped, so update the encoded frame size
// (encoder callback is only called for non-zero length frames).
encoded_frame_size_ = encoded_image->_length;
TickTime encode_stop = TickTime::Now();
int frame_number = encoded_image->_timeStamp;
FrameStatistic& stat = stats_->stats_[frame_number];
stat.encode_time_in_us = GetElapsedTimeMicroseconds(encode_start_,
encode_stop);
stat.encoding_successful = true;
stat.encoded_frame_length_in_bytes = encoded_image->_length;
stat.frame_number = encoded_image->_timeStamp;
stat.frame_type = encoded_image->_frameType;
stat.bit_rate_in_kbps = encoded_image->_length * bit_rate_factor_;
stat.total_packets = encoded_image->_length /
config_.networking_config.packet_size_in_bytes + 1;
// Perform packet loss if criteria is fullfilled:
bool exclude_this_frame = false;
// Only keyframes can be excluded
if (encoded_image->_frameType == kKeyFrame) {
switch (config_.exclude_frame_types) {
case kExcludeOnlyFirstKeyFrame:
if (!first_key_frame_has_been_excluded_) {
first_key_frame_has_been_excluded_ = true;
exclude_this_frame = true;
}
break;
case kExcludeAllKeyFrames:
exclude_this_frame = true;
break;
default:
assert(false);
}
}
if (!exclude_this_frame) {
stat.packets_dropped =
packet_manipulator_->ManipulatePackets(encoded_image);
}
// Keep track of if frames are lost due to packet loss so we can tell
// this to the encoder (this is handled by the RTP logic in the full stack)
decode_start_ = TickTime::Now();
// TODO(kjellander): Pass fragmentation header to the decoder when
// CL 172001 has been submitted and PacketManipulator supports this.
WebRtc_Word32 decode_result = decoder_->Decode(*encoded_image,
last_frame_missing_, NULL);
stat.decode_return_code = decode_result;
if (decode_result != WEBRTC_VIDEO_CODEC_OK) {
// Write the last successful frame the output file to avoid getting it out
// of sync with the source file for SSIM and PSNR comparisons:
frame_writer_->WriteFrame(last_successful_frame_buffer_);
}
// save status for losses so we can inform the decoder for the next frame:
last_frame_missing_ = encoded_image->_length == 0;
}
void VideoProcessorImpl::FrameDecoded(const VideoFrame& image) {
TickTime decode_stop = TickTime::Now();
int frame_number = image.TimeStamp();
// Report stats
FrameStatistic& stat = stats_->stats_[frame_number];
stat.decode_time_in_us = GetElapsedTimeMicroseconds(decode_start_,
decode_stop);
stat.decoding_successful = true;
// Check for resize action (either down or up):
if (static_cast<int>(image.Width()) != last_encoder_frame_width_ ||
static_cast<int>(image.Height()) != last_encoder_frame_height_ ) {
++num_spatial_resizes_;
last_encoder_frame_width_ = image.Width();
last_encoder_frame_height_ = image.Height();
}
// Check if codec size is different from native/original size, and if so,
// upsample back to original size: needed for PSNR and SSIM computations.
if (image.Width() != config_.codec_settings->width ||
image.Height() != config_.codec_settings->height) {
VideoFrame up_image;
int ret_val = scaler_.Set(image.Width(), image.Height(),
config_.codec_settings->width,
config_.codec_settings->height,
kI420, kI420, kScaleBilinear);
assert(ret_val >= 0);
if (ret_val < 0) {
fprintf(stderr, "Failed to set scalar for frame: %d, return code: %d\n",
frame_number, ret_val);
}
ret_val = scaler_.Scale(image, &up_image);
assert(ret_val >= 0);
if (ret_val < 0) {
fprintf(stderr, "Failed to scale frame: %d, return code: %d\n",
frame_number, ret_val);
}
// Update our copy of the last successful frame:
memcpy(last_successful_frame_buffer_, up_image.Buffer(), up_image.Length());
bool write_success = frame_writer_->WriteFrame(up_image.Buffer());
assert(write_success);
if (!write_success) {
fprintf(stderr, "Failed to write frame %d to disk!", frame_number);
}
up_image.Free();
} else { // No resize.
// Update our copy of the last successful frame:
memcpy(last_successful_frame_buffer_, image.Buffer(), image.Length());
bool write_success = frame_writer_->WriteFrame(image.Buffer());
assert(write_success);
if (!write_success) {
fprintf(stderr, "Failed to write frame %d to disk!", frame_number);
}
}
}
int VideoProcessorImpl::GetElapsedTimeMicroseconds(
const webrtc::TickTime& start, const webrtc::TickTime& stop) {
WebRtc_UWord64 encode_time = (stop - start).Microseconds();
assert(encode_time <
static_cast<unsigned int>(std::numeric_limits<int>::max()));
return static_cast<int>(encode_time);
}
const char* ExcludeFrameTypesToStr(ExcludeFrameTypes e) {
switch (e) {
case kExcludeOnlyFirstKeyFrame:
return "ExcludeOnlyFirstKeyFrame";
case kExcludeAllKeyFrames:
return "ExcludeAllKeyFrames";
default:
assert(false);
return "Unknown";
}
}
const char* VideoCodecTypeToStr(webrtc::VideoCodecType e) {
switch (e) {
case kVideoCodecVP8:
return "VP8";
case kVideoCodecI420:
return "I420";
case kVideoCodecRED:
return "RED";
case kVideoCodecULPFEC:
return "ULPFEC";
case kVideoCodecUnknown:
return "Unknown";
default:
assert(false);
return "Unknown";
}
}
// Callbacks
WebRtc_Word32
VideoProcessorImpl::VideoProcessorEncodeCompleteCallback::Encoded(
EncodedImage& encoded_image,
const webrtc::CodecSpecificInfo* codec_specific_info,
const webrtc::RTPFragmentationHeader* fragmentation) {
video_processor_->FrameEncoded(&encoded_image); // Forward to parent class.
return 0;
}
WebRtc_Word32
VideoProcessorImpl::VideoProcessorDecodeCompleteCallback::Decoded(
VideoFrame& image) {
video_processor_->FrameDecoded(image); // forward to parent class
return 0;
}
} // namespace test
} // namespace webrtc

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_VIDEOPROCESSOR_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_VIDEOPROCESSOR_H_
#include <string>
#include "common_video/libyuv/include/webrtc_libyuv.h"
#include "common_video/libyuv/include/scaler.h"
#include "modules/interface/module_common_types.h"
#include "modules/video_coding/codecs/interface/video_codec_interface.h"
#include "modules/video_coding/codecs/test/packet_manipulator.h"
#include "modules/video_coding/codecs/test/stats.h"
#include "system_wrappers/interface/tick_util.h"
#include "testsupport/frame_reader.h"
#include "testsupport/frame_writer.h"
namespace webrtc {
namespace test {
// Defines which frame types shall be excluded from packet loss and when.
enum ExcludeFrameTypes {
// Will exclude the first keyframe in the video sequence from packet loss.
// Following keyframes will be targeted for packet loss.
kExcludeOnlyFirstKeyFrame,
// Exclude all keyframes from packet loss, no matter where in the video
// sequence they occur.
kExcludeAllKeyFrames
};
// Returns a string representation of the enum value.
const char* ExcludeFrameTypesToStr(ExcludeFrameTypes e);
// Test configuration for a test run
struct TestConfig {
TestConfig()
: name(""), description(""), test_number(0),
input_filename(""), output_filename(""), output_dir("out"),
networking_config(), exclude_frame_types(kExcludeOnlyFirstKeyFrame),
frame_length_in_bytes(-1), use_single_core(false), keyframe_interval(0),
codec_settings(NULL), verbose(true) {
};
// Name of the test. This is purely metadata and does not affect
// the test in any way.
std::string name;
// More detailed description of the test. This is purely metadata and does
// not affect the test in any way.
std::string description;
// Number of this test. Useful if multiple runs of the same test with
// different configurations shall be managed.
int test_number;
// File to process for the test. This must be a video file in the YUV format.
std::string input_filename;
// File to write to during processing for the test. Will be a video file
// in the YUV format.
std::string output_filename;
// Path to the directory where encoded files will be put
// (absolute or relative to the executable). Default: "out".
std::string output_dir;
// Configurations related to networking.
NetworkingConfig networking_config;
// Decides how the packet loss simulations shall exclude certain frames
// from packet loss. Default: kExcludeOnlyFirstKeyFrame.
ExcludeFrameTypes exclude_frame_types;
// The length of a single frame of the input video file. This value is
// calculated out of the width and height according to the video format
// specification. Must be set before processing.
int frame_length_in_bytes;
// Force the encoder and decoder to use a single core for processing.
// Using a single core is necessary to get a deterministic behavior for the
// encoded frames - using multiple cores will produce different encoded frames
// since multiple cores are competing to consume the byte budget for each
// frame in parallel.
// If set to false, the maximum number of available cores will be used.
// Default: false.
bool use_single_core;
// If set to a value >0 this setting forces the encoder to create a keyframe
// every Nth frame. Note that the encoder may create a keyframe in other
// locations in addition to the interval that is set using this parameter.
// Forcing key frames may also affect encoder planning optimizations in
// a negative way, since it will suddenly be forced to produce an expensive
// key frame.
// Default: 0.
int keyframe_interval;
// The codec settings to use for the test (target bitrate, video size,
// framerate and so on). This struct must be created and filled in using
// the VideoCodingModule::Codec() method.
webrtc::VideoCodec* codec_settings;
// If printing of information to stdout shall be performed during processing.
bool verbose;
};
// Returns a string representation of the enum value.
const char* VideoCodecTypeToStr(webrtc::VideoCodecType e);
// Handles encoding/decoding of video using the VideoEncoder/VideoDecoder
// interfaces. This is done in a sequential manner in order to be able to
// measure times properly.
// The class processes a frame at the time for the configured input file.
// It maintains state of where in the source input file the processing is at.
//
// Regarding packet loss: Note that keyframes are excluded (first or all
// depending on the ExcludeFrameTypes setting). This is because if key frames
// would be altered, all the following delta frames would be pretty much
// worthless. VP8 has an error-resilience feature that makes it able to handle
// packet loss in key non-first keyframes, which is why only the first is
// excluded by default.
// Packet loss in such important frames is handled on a higher level in the
// Video Engine, where signaling would request a retransmit of the lost packets,
// since they're so important.
//
// Note this class is not thread safe in any way and is meant for simple testing
// purposes.
class VideoProcessor {
public:
virtual ~VideoProcessor() {}
// Performs initial calculations about frame size, sets up callbacks etc.
// Returns false if an error has occurred, in addition to printing to stderr.
virtual bool Init() = 0;
// Processes a single frame. Returns true as long as there's more frames
// available in the source clip.
// Frame number must be an integer >=0.
virtual bool ProcessFrame(int frame_number) = 0;
// Updates the encoder with the target bit rate and the frame rate.
virtual void SetRates(int bit_rate, int frame_rate) = 0;
// Return the size of the encoded frame in bytes. Dropped frames by the
// encoder are regarded as zero size.
virtual int EncodedFrameSize() = 0;
// Return the number of dropped frames.
virtual int NumberDroppedFrames() = 0;
// Return the number of spatial resizes.
virtual int NumberSpatialResizes() = 0;
};
class VideoProcessorImpl : public VideoProcessor {
public:
VideoProcessorImpl(webrtc::VideoEncoder* encoder,
webrtc::VideoDecoder* decoder,
FrameReader* frame_reader,
FrameWriter* frame_writer,
PacketManipulator* packet_manipulator,
const TestConfig& config,
Stats* stats);
virtual ~VideoProcessorImpl();
virtual bool Init();
virtual bool ProcessFrame(int frame_number);
private:
// Invoked by the callback when a frame has completed encoding.
void FrameEncoded(webrtc::EncodedImage* encodedImage);
// Invoked by the callback when a frame has completed decoding.
void FrameDecoded(const webrtc::VideoFrame& image);
// Used for getting a 32-bit integer representing time
// (checks the size is within signed 32-bit bounds before casting it)
int GetElapsedTimeMicroseconds(const webrtc::TickTime& start,
const webrtc::TickTime& stop);
// Updates the encoder with the target bit rate and the frame rate.
void SetRates(int bit_rate, int frame_rate);
// Return the size of the encoded frame in bytes.
int EncodedFrameSize();
// Return the number of dropped frames.
int NumberDroppedFrames();
// Return the number of spatial resizes.
int NumberSpatialResizes();
webrtc::VideoEncoder* encoder_;
webrtc::VideoDecoder* decoder_;
FrameReader* frame_reader_;
FrameWriter* frame_writer_;
PacketManipulator* packet_manipulator_;
const TestConfig& config_;
Stats* stats_;
EncodedImageCallback* encode_callback_;
DecodedImageCallback* decode_callback_;
// Buffer used for reading the source video file:
WebRtc_UWord8* source_buffer_;
// Keep track of the last successful frame, since we need to write that
// when decoding fails:
WebRtc_UWord8* last_successful_frame_buffer_;
webrtc::VideoFrame source_frame_;
// To keep track of if we have excluded the first key frame from packet loss:
bool first_key_frame_has_been_excluded_;
// To tell the decoder previous frame have been dropped due to packet loss:
bool last_frame_missing_;
// If Init() has executed successfully.
bool initialized_;
int encoded_frame_size_;
int prev_time_stamp_;
int num_dropped_frames_;
int num_spatial_resizes_;
int last_encoder_frame_width_;
int last_encoder_frame_height_;
Scaler scaler_;
// Statistics
double bit_rate_factor_; // multiply frame length with this to get bit rate
webrtc::TickTime encode_start_;
webrtc::TickTime decode_start_;
// Callback class required to implement according to the VideoEncoder API.
class VideoProcessorEncodeCompleteCallback
: public webrtc::EncodedImageCallback {
public:
explicit VideoProcessorEncodeCompleteCallback(VideoProcessorImpl* vp)
: video_processor_(vp) {
}
WebRtc_Word32 Encoded(
webrtc::EncodedImage& encoded_image,
const webrtc::CodecSpecificInfo* codec_specific_info = NULL,
const webrtc::RTPFragmentationHeader* fragmentation = NULL);
private:
VideoProcessorImpl* video_processor_;
};
// Callback class required to implement according to the VideoDecoder API.
class VideoProcessorDecodeCompleteCallback
: public webrtc::DecodedImageCallback {
public:
explicit VideoProcessorDecodeCompleteCallback(VideoProcessorImpl* vp)
: video_processor_(vp) {
}
WebRtc_Word32 Decoded(webrtc::VideoFrame& image);
private:
VideoProcessorImpl* video_processor_;
};
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_VIDEOPROCESSOR_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "gtest/gtest.h"
#include <math.h>
#include "modules/video_coding/codecs/interface/video_codec_interface.h"
#include "modules/video_coding/codecs/test/packet_manipulator.h"
#include "modules/video_coding/codecs/test/videoprocessor.h"
#include "modules/video_coding/codecs/vp8/include/vp8.h"
#include "modules/video_coding/codecs/vp8/include/vp8_common_types.h"
#include "modules/video_coding/main/interface/video_coding.h"
#include "testsupport/fileutils.h"
#include "testsupport/frame_reader.h"
#include "testsupport/frame_writer.h"
#include "testsupport/metrics/video_metrics.h"
#include "testsupport/packet_reader.h"
#include "typedefs.h"
namespace webrtc {
// Maximum number of rate updates (i.e., calls to encoder to change bitrate
// and/or frame rate) for the current tests.
const int kMaxNumRateUpdates = 3;
const int kPercTargetvsActualMismatch = 20;
// Codec and network settings.
struct CodecConfigPars {
float packet_loss;
int num_temporal_layers;
int key_frame_interval;
bool error_concealment_on;
bool denoising_on;
bool frame_dropper_on;
bool spatial_resize_on;
};
// Quality metrics.
struct QualityMetrics {
double minimum_avg_psnr;
double minimum_min_psnr;
double minimum_avg_ssim;
double minimum_min_ssim;
};
// The sequence of bitrate and frame rate changes for the encoder, the frame
// number where the changes are made, and the total number of frames for the
// test.
struct RateProfile {
int target_bit_rate[kMaxNumRateUpdates];
int input_frame_rate[kMaxNumRateUpdates];
int frame_index_rate_update[kMaxNumRateUpdates + 1];
int num_frames;
};
// Metrics for the rate control. The rate mismatch metrics are defined as
// percentages.|max_time_hit_target| is defined as number of frames, after a
// rate update is made to the encoder, for the encoder to reach within
// |kPercTargetvsActualMismatch| of new target rate. The metrics are defined for
// each rate update sequence.
struct RateControlMetrics {
int max_num_dropped_frames;
int max_key_frame_size_mismatch;
int max_delta_frame_size_mismatch;
int max_encoding_rate_mismatch;
int max_time_hit_target;
int num_spatial_resizes;
};
// Sequence used is foreman (CIF): may be better to use VGA for resize test.
const int kCIFWidth = 352;
const int kCIFHeight = 288;
const int kNbrFramesShort = 100; // Some tests are run for shorter sequence.
const int kNbrFramesLong = 299;
// Parameters from VP8 wrapper, which control target size of key frames.
const float kInitialBufferSize = 0.5f;
const float kOptimalBufferSize = 0.6f;
const float kScaleKeyFrameSize = 0.5f;
// Integration test for video processor. Encodes+decodes a clip and
// writes it to the output directory. After completion, quality metrics
// (PSNR and SSIM) and rate control metrics are computed to verify that the
// quality and encoder response is acceptable. The rate control tests allow us
// to verify the behavior for changing bitrate, changing frame rate, frame
// dropping/spatial resize, and temporal layers. The limits for the rate
// control metrics are set to be fairly conservative, so failure should only
// happen when some significant regression or breakdown occurs.
class VideoProcessorIntegrationTest: public testing::Test {
protected:
VideoEncoder* encoder_;
VideoDecoder* decoder_;
webrtc::test::FrameReader* frame_reader_;
webrtc::test::FrameWriter* frame_writer_;
webrtc::test::PacketReader packet_reader_;
webrtc::test::PacketManipulator* packet_manipulator_;
webrtc::test::Stats stats_;
webrtc::test::TestConfig config_;
VideoCodec codec_settings_;
webrtc::test::VideoProcessor* processor_;
// Quantities defined/updated for every encoder rate update.
// Some quantities defined per temporal layer (at most 3 layers in this test).
int num_frames_per_update_[3];
float sum_frame_size_mismatch_[3];
float sum_encoded_frame_size_[3];
float encoding_bitrate_[3];
float per_frame_bandwidth_[3];
float bit_rate_layer_[3];
float frame_rate_layer_[3];
int num_frames_total_;
float sum_encoded_frame_size_total_;
float encoding_bitrate_total_;
float perc_encoding_rate_mismatch_;
int num_frames_to_hit_target_;
bool encoding_rate_within_target_;
int bit_rate_;
int frame_rate_;
int layer_;
float target_size_key_frame_initial_;
float target_size_key_frame_;
float sum_key_frame_size_mismatch_;
int num_key_frames_;
float start_bitrate_;
// Codec and network settings.
float packet_loss_;
int num_temporal_layers_;
int key_frame_interval_;
bool error_concealment_on_;
bool denoising_on_;
bool frame_dropper_on_;
bool spatial_resize_on_;
VideoProcessorIntegrationTest() {}
virtual ~VideoProcessorIntegrationTest() {}
void SetUpCodecConfig() {
encoder_ = VP8Encoder::Create();
decoder_ = VP8Decoder::Create();
// CIF is currently used for all tests below.
// Setup the TestConfig struct for processing of a clip in CIF resolution.
config_.input_filename =
webrtc::test::ResourcePath("foreman_cif", "yuv");
config_.output_filename = webrtc::test::OutputPath() +
"foreman_cif_short_video_codecs_test_framework_integrationtests.yuv";
config_.frame_length_in_bytes = 3 * kCIFWidth * kCIFHeight / 2;
config_.verbose = false;
// Only allow encoder/decoder to use single core, for predictability.
config_.use_single_core = true;
// Key frame interval and packet loss are set for each test.
config_.keyframe_interval = key_frame_interval_;
config_.networking_config.packet_loss_probability = packet_loss_;
// Get a codec configuration struct and configure it.
VideoCodingModule::Codec(kVideoCodecVP8, &codec_settings_);
config_.codec_settings = &codec_settings_;
config_.codec_settings->startBitrate = start_bitrate_;
config_.codec_settings->width = kCIFWidth;
config_.codec_settings->height = kCIFHeight;
// These features may be set depending on the test.
config_.codec_settings->codecSpecific.VP8.errorConcealmentOn =
error_concealment_on_;
config_.codec_settings->codecSpecific.VP8.denoisingOn =
denoising_on_;
config_.codec_settings->codecSpecific.VP8.numberOfTemporalLayers =
num_temporal_layers_;
config_.codec_settings->codecSpecific.VP8.frameDroppingOn =
frame_dropper_on_;
config_.codec_settings->codecSpecific.VP8.automaticResizeOn =
spatial_resize_on_;
frame_reader_ =
new webrtc::test::FrameReaderImpl(config_.input_filename,
config_.frame_length_in_bytes);
frame_writer_ =
new webrtc::test::FrameWriterImpl(config_.output_filename,
config_.frame_length_in_bytes);
ASSERT_TRUE(frame_reader_->Init());
ASSERT_TRUE(frame_writer_->Init());
packet_manipulator_ = new webrtc::test::PacketManipulatorImpl(
&packet_reader_, config_.networking_config, config_.verbose);
processor_ = new webrtc::test::VideoProcessorImpl(encoder_, decoder_,
frame_reader_,
frame_writer_,
packet_manipulator_,
config_, &stats_);
ASSERT_TRUE(processor_->Init());
}
// Reset quantities after each encoder update, update the target
// per-frame bandwidth.
void ResetRateControlMetrics(int num_frames) {
for (int i = 0; i < num_temporal_layers_; i++) {
num_frames_per_update_[i] = 0;
sum_frame_size_mismatch_[i] = 0.0f;
sum_encoded_frame_size_[i] = 0.0f;
encoding_bitrate_[i] = 0.0f;
// Update layer per-frame-bandwidth.
per_frame_bandwidth_[i] = static_cast<float>(bit_rate_layer_[i]) /
static_cast<float>(frame_rate_layer_[i]);
}
// Set maximum size of key frames, following setting in the VP8 wrapper.
float max_key_size = kScaleKeyFrameSize * kOptimalBufferSize * frame_rate_;
// We don't know exact target size of the key frames (except for first one),
// but the minimum in libvpx is ~|3 * per_frame_bandwidth| and maximum is
// set by |max_key_size_ * per_frame_bandwidth|. Take middle point/average
// as reference for mismatch. Note key frames always correspond to base
// layer frame in this test.
target_size_key_frame_ = 0.5 * (3 + max_key_size) * per_frame_bandwidth_[0];
num_frames_total_ = 0;
sum_encoded_frame_size_total_ = 0.0f;
encoding_bitrate_total_ = 0.0f;
perc_encoding_rate_mismatch_ = 0.0f;
num_frames_to_hit_target_ = num_frames;
encoding_rate_within_target_ = false;
sum_key_frame_size_mismatch_ = 0.0;
num_key_frames_ = 0;
}
// For every encoded frame, update the rate control metrics.
void UpdateRateControlMetrics(int frame_num, VideoFrameType frame_type) {
int encoded_frame_size = processor_->EncodedFrameSize();
float encoded_size_kbits = encoded_frame_size * 8.0f / 1000.0f;
// Update layer data.
// Update rate mismatch relative to per-frame bandwidth for delta frames.
if (frame_type == kDeltaFrame) {
// TODO(marpan): Should we count dropped (zero size) frames in mismatch?
sum_frame_size_mismatch_[layer_] += fabs(encoded_size_kbits -
per_frame_bandwidth_[layer_]) /
per_frame_bandwidth_[layer_];
} else {
float target_size = (frame_num == 1) ? target_size_key_frame_initial_ :
target_size_key_frame_;
sum_key_frame_size_mismatch_ += fabs(encoded_size_kbits - target_size) /
target_size;
num_key_frames_ += 1;
}
sum_encoded_frame_size_[layer_] += encoded_size_kbits;
// Encoding bitrate per layer: from the start of the update/run to the
// current frame.
encoding_bitrate_[layer_] = sum_encoded_frame_size_[layer_] *
frame_rate_layer_[layer_] /
num_frames_per_update_[layer_];
// Total encoding rate: from the start of the update/run to current frame.
sum_encoded_frame_size_total_ += encoded_size_kbits;
encoding_bitrate_total_ = sum_encoded_frame_size_total_ * frame_rate_ /
num_frames_total_;
perc_encoding_rate_mismatch_ = 100 * fabs(encoding_bitrate_total_ -
bit_rate_) / bit_rate_;
if (perc_encoding_rate_mismatch_ < kPercTargetvsActualMismatch &&
!encoding_rate_within_target_) {
num_frames_to_hit_target_ = num_frames_total_;
encoding_rate_within_target_ = true;
}
}
// Verify expected behavior of rate control and print out data.
void VerifyRateControl(int update_index,
int max_key_frame_size_mismatch,
int max_delta_frame_size_mismatch,
int max_encoding_rate_mismatch,
int max_time_hit_target,
int max_num_dropped_frames,
int num_spatial_resizes) {
int num_dropped_frames = processor_->NumberDroppedFrames();
int num_resize_actions = processor_->NumberSpatialResizes();
printf("For update #: %d,\n "
" Target Bitrate: %d,\n"
" Encoding bitrate: %f,\n"
" Frame rate: %d \n",
update_index, bit_rate_, encoding_bitrate_total_, frame_rate_);
printf(" Number of frames to approach target rate = %d, \n"
" Number of dropped frames = %d, \n"
" Number of spatial resizes = %d, \n",
num_frames_to_hit_target_, num_dropped_frames, num_resize_actions);
EXPECT_LE(perc_encoding_rate_mismatch_, max_encoding_rate_mismatch);
if (num_key_frames_ > 0) {
int perc_key_frame_size_mismatch = 100 * sum_key_frame_size_mismatch_ /
num_key_frames_;
printf(" Number of Key frames: %d \n"
" Key frame rate mismatch: %d \n",
num_key_frames_, perc_key_frame_size_mismatch);
EXPECT_LE(perc_key_frame_size_mismatch, max_key_frame_size_mismatch);
}
printf("\n");
printf("Rates statistics for Layer data \n");
for (int i = 0; i < num_temporal_layers_ ; i++) {
printf("Layer #%d \n", i);
int perc_frame_size_mismatch = 100 * sum_frame_size_mismatch_[i] /
num_frames_per_update_[i];
int perc_encoding_rate_mismatch = 100 * fabs(encoding_bitrate_[i] -
bit_rate_layer_[i]) /
bit_rate_layer_[i];
printf(" Target Layer Bit rate: %f \n"
" Layer frame rate: %f, \n"
" Layer per frame bandwidth: %f, \n"
" Layer Encoding bit rate: %f, \n"
" Layer Percent frame size mismatch: %d, \n"
" Layer Percent encoding rate mismatch = %d, \n"
" Number of frame processed per layer = %d \n",
bit_rate_layer_[i], frame_rate_layer_[i], per_frame_bandwidth_[i],
encoding_bitrate_[i], perc_frame_size_mismatch,
perc_encoding_rate_mismatch, num_frames_per_update_[i]);
EXPECT_LE(perc_frame_size_mismatch, max_delta_frame_size_mismatch);
EXPECT_LE(perc_encoding_rate_mismatch, max_encoding_rate_mismatch);
}
printf("\n");
EXPECT_LE(num_frames_to_hit_target_, max_time_hit_target);
EXPECT_LE(num_dropped_frames, max_num_dropped_frames);
EXPECT_EQ(num_resize_actions, num_spatial_resizes);
}
// Layer index corresponding to frame number, for up to 3 layers.
void LayerIndexForFrame(int frame_number) {
if (num_temporal_layers_ == 1) {
layer_ = 0;
} else if (num_temporal_layers_ == 2) {
// layer 0: 0 2 4 ...
// layer 1: 1 3
if (frame_number % 2 == 0) {
layer_ = 0;
} else {
layer_ = 1;
}
} else if (num_temporal_layers_ == 3) {
// layer 0: 0 4 8 ...
// layer 1: 2 6
// layer 2: 1 3 5 7
if (frame_number % 4 == 0) {
layer_ = 0;
} else if ((frame_number + 2) % 4 == 0) {
layer_ = 1;
} else if ((frame_number + 1) % 2 == 0) {
layer_ = 2;
}
} else {
assert(false); // Only up to 3 layers.
}
}
// Set the bitrate and frame rate per layer, for up to 3 layers.
void SetLayerRates() {
assert(num_temporal_layers_<= 3);
for (int i = 0; i < num_temporal_layers_; i++) {
float bit_rate_ratio =
kVp8LayerRateAlloction[num_temporal_layers_ - 1][i];
if (i > 0) {
float bit_rate_delta_ratio = kVp8LayerRateAlloction
[num_temporal_layers_ - 1][i] -
kVp8LayerRateAlloction[num_temporal_layers_ - 1][i - 1];
bit_rate_layer_[i] = bit_rate_ * bit_rate_delta_ratio;
} else {
bit_rate_layer_[i] = bit_rate_ * bit_rate_ratio;
}
frame_rate_layer_[i] = frame_rate_ / static_cast<float>(
1 << (num_temporal_layers_ - 1));
}
if (num_temporal_layers_ == 3) {
frame_rate_layer_[2] = frame_rate_ / 2.0f;
}
}
VideoFrameType FrameType(int frame_number) {
if (frame_number == 0 || ((frame_number) % key_frame_interval_ == 0 &&
key_frame_interval_ > 0)) {
return kKeyFrame;
} else {
return kDeltaFrame;
}
}
void TearDown() {
delete processor_;
delete packet_manipulator_;
delete frame_writer_;
delete frame_reader_;
delete decoder_;
delete encoder_;
}
// Processes all frames in the clip and verifies the result.
void ProcessFramesAndVerify(QualityMetrics quality_metrics,
RateProfile rate_profile,
CodecConfigPars process,
RateControlMetrics* rc_metrics) {
// Codec/config settings.
start_bitrate_ = rate_profile.target_bit_rate[0];
packet_loss_ = process.packet_loss;
key_frame_interval_ = process.key_frame_interval;
num_temporal_layers_ = process.num_temporal_layers;
error_concealment_on_ = process.error_concealment_on;
denoising_on_ = process.denoising_on;
frame_dropper_on_ = process.frame_dropper_on;
spatial_resize_on_ = process.spatial_resize_on;
SetUpCodecConfig();
// Update the layers and the codec with the initial rates.
bit_rate_ = rate_profile.target_bit_rate[0];
frame_rate_ = rate_profile.input_frame_rate[0];
SetLayerRates();
// Set the initial target size for key frame.
target_size_key_frame_initial_ = 0.5 * kInitialBufferSize *
bit_rate_layer_[0];
processor_->SetRates(bit_rate_, frame_rate_);
// Process each frame, up to |num_frames|.
int num_frames = rate_profile.num_frames;
int update_index = 0;
ResetRateControlMetrics(
rate_profile.frame_index_rate_update[update_index + 1]);
int frame_number = 0;
VideoFrameType frame_type = kDeltaFrame;
while (processor_->ProcessFrame(frame_number) &&
frame_number < num_frames) {
// Get the layer index for the frame |frame_number|.
LayerIndexForFrame(frame_number);
frame_type = FrameType(frame_number);
// Counter for whole sequence run.
++frame_number;
// Counters for each rate update.
++num_frames_per_update_[layer_];
++num_frames_total_;
UpdateRateControlMetrics(frame_number, frame_type);
// If we hit another/next update, verify stats for current state and
// update layers and codec with new rates.
if (frame_number ==
rate_profile.frame_index_rate_update[update_index + 1]) {
VerifyRateControl(
update_index,
rc_metrics[update_index].max_key_frame_size_mismatch,
rc_metrics[update_index].max_delta_frame_size_mismatch,
rc_metrics[update_index].max_encoding_rate_mismatch,
rc_metrics[update_index].max_time_hit_target,
rc_metrics[update_index].max_num_dropped_frames,
rc_metrics[update_index].num_spatial_resizes);
// Update layer rates and the codec with new rates.
++update_index;
bit_rate_ = rate_profile.target_bit_rate[update_index];
frame_rate_ = rate_profile.input_frame_rate[update_index];
SetLayerRates();
ResetRateControlMetrics(rate_profile.
frame_index_rate_update[update_index + 1]);
processor_->SetRates(bit_rate_, frame_rate_);
}
}
VerifyRateControl(
update_index,
rc_metrics[update_index].max_key_frame_size_mismatch,
rc_metrics[update_index].max_delta_frame_size_mismatch,
rc_metrics[update_index].max_encoding_rate_mismatch,
rc_metrics[update_index].max_time_hit_target,
rc_metrics[update_index].max_num_dropped_frames,
rc_metrics[update_index].num_spatial_resizes);
EXPECT_EQ(num_frames, frame_number);
EXPECT_EQ(num_frames + 1, static_cast<int>(stats_.stats_.size()));
// Release encoder and decoder to make sure they have finished processing:
EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Release());
// Close the files before we start using them for SSIM/PSNR calculations.
frame_reader_->Close();
frame_writer_->Close();
// TODO(marpan): should compute these quality metrics per SetRates update.
webrtc::test::QualityMetricsResult psnr_result, ssim_result;
EXPECT_EQ(0, webrtc::test::I420MetricsFromFiles(
config_.input_filename.c_str(),
config_.output_filename.c_str(),
config_.codec_settings->width,
config_.codec_settings->height,
&psnr_result,
&ssim_result));
printf("PSNR avg: %f, min: %f SSIM avg: %f, min: %f\n",
psnr_result.average, psnr_result.min,
ssim_result.average, ssim_result.min);
stats_.PrintSummary();
EXPECT_GT(psnr_result.average, quality_metrics.minimum_avg_psnr);
EXPECT_GT(psnr_result.min, quality_metrics.minimum_min_psnr);
EXPECT_GT(ssim_result.average, quality_metrics.minimum_avg_ssim);
EXPECT_GT(ssim_result.min, quality_metrics.minimum_min_ssim);
}
};
void SetRateProfilePars(RateProfile* rate_profile,
int update_index,
int bit_rate,
int frame_rate,
int frame_index_rate_update) {
rate_profile->target_bit_rate[update_index] = bit_rate;
rate_profile->input_frame_rate[update_index] = frame_rate;
rate_profile->frame_index_rate_update[update_index] = frame_index_rate_update;
}
void SetCodecParameters(CodecConfigPars* process_settings,
float packet_loss,
int key_frame_interval,
int num_temporal_layers,
bool error_concealment_on,
bool denoising_on,
bool frame_dropper_on,
bool spatial_resize_on) {
process_settings->packet_loss = packet_loss;
process_settings->key_frame_interval = key_frame_interval;
process_settings->num_temporal_layers = num_temporal_layers,
process_settings->error_concealment_on = error_concealment_on;
process_settings->denoising_on = denoising_on;
process_settings->frame_dropper_on = frame_dropper_on;
process_settings->spatial_resize_on = spatial_resize_on;
}
void SetQualityMetrics(QualityMetrics* quality_metrics,
double minimum_avg_psnr,
double minimum_min_psnr,
double minimum_avg_ssim,
double minimum_min_ssim) {
quality_metrics->minimum_avg_psnr = minimum_avg_psnr;
quality_metrics->minimum_min_psnr = minimum_min_psnr;
quality_metrics->minimum_avg_ssim = minimum_avg_ssim;
quality_metrics->minimum_min_ssim = minimum_min_ssim;
}
void SetRateControlMetrics(RateControlMetrics* rc_metrics,
int update_index,
int max_num_dropped_frames,
int max_key_frame_size_mismatch,
int max_delta_frame_size_mismatch,
int max_encoding_rate_mismatch,
int max_time_hit_target,
int num_spatial_resizes) {
rc_metrics[update_index].max_num_dropped_frames = max_num_dropped_frames;
rc_metrics[update_index].max_key_frame_size_mismatch =
max_key_frame_size_mismatch;
rc_metrics[update_index].max_delta_frame_size_mismatch =
max_delta_frame_size_mismatch;
rc_metrics[update_index].max_encoding_rate_mismatch =
max_encoding_rate_mismatch;
rc_metrics[update_index].max_time_hit_target = max_time_hit_target;
rc_metrics[update_index].num_spatial_resizes = num_spatial_resizes;
}
// Run with no packet loss and fixed bitrate. Quality should be very high.
// One key frame (first frame only) in sequence. Setting |key_frame_interval|
// to -1 below means no periodic key frames in test.
TEST_F(VideoProcessorIntegrationTest, ProcessZeroPacketLoss) {
// Bitrate and frame rate profile.
RateProfile rate_profile;
SetRateProfilePars(&rate_profile, 0, 500, 30, 0);
rate_profile.frame_index_rate_update[1] = kNbrFramesShort + 1;
rate_profile.num_frames = kNbrFramesShort;
// Codec/network settings.
CodecConfigPars process_settings;
SetCodecParameters(&process_settings, 0.0f, -1, 1, true, true, true, false);
// Metrics for expected quality.
QualityMetrics quality_metrics;
SetQualityMetrics(&quality_metrics, 36.95, 33.0, 0.90, 0.90);
// Metrics for rate control.
RateControlMetrics rc_metrics[1];
SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 15, 0);
ProcessFramesAndVerify(quality_metrics,
rate_profile,
process_settings,
rc_metrics);
}
// Run with 5% packet loss and fixed bitrate. Quality should be a bit lower.
// One key frame (first frame only) in sequence.
TEST_F(VideoProcessorIntegrationTest, Process5PercentPacketLoss) {
// Bitrate and frame rate profile.
RateProfile rate_profile;
SetRateProfilePars(&rate_profile, 0, 500, 30, 0);
rate_profile.frame_index_rate_update[1] = kNbrFramesShort + 1;
rate_profile.num_frames = kNbrFramesShort;
// Codec/network settings.
CodecConfigPars process_settings;
SetCodecParameters(&process_settings, 0.05f, -1, 1, true, true, true, false);
// Metrics for expected quality.
QualityMetrics quality_metrics;
SetQualityMetrics(&quality_metrics, 20.0, 16.0, 0.60, 0.40);
// Metrics for rate control.
RateControlMetrics rc_metrics[1];
SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 15, 0);
ProcessFramesAndVerify(quality_metrics,
rate_profile,
process_settings,
rc_metrics);
}
// Run with 10% packet loss and fixed bitrate. Quality should be even lower.
// One key frame (first frame only) in sequence.
TEST_F(VideoProcessorIntegrationTest, Process10PercentPacketLoss) {
// Bitrate and frame rate profile.
RateProfile rate_profile;
SetRateProfilePars(&rate_profile, 0, 500, 30, 0);
rate_profile.frame_index_rate_update[1] = kNbrFramesShort + 1;
rate_profile.num_frames = kNbrFramesShort;
// Codec/network settings.
CodecConfigPars process_settings;
SetCodecParameters(&process_settings, 0.1f, -1, 1, true, true, true, false);
// Metrics for expected quality.
QualityMetrics quality_metrics;
SetQualityMetrics(&quality_metrics, 19.0, 16.0, 0.50, 0.35);
// Metrics for rate control.
RateControlMetrics rc_metrics[1];
SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 15, 0);
ProcessFramesAndVerify(quality_metrics,
rate_profile,
process_settings,
rc_metrics);
}
// Run with no packet loss, with varying bitrate (3 rate updates):
// low to high to medium. Check that quality and encoder response to the new
// target rate/per-frame bandwidth (for each rate update) is within limits.
// One key frame (first frame only) in sequence.
TEST_F(VideoProcessorIntegrationTest, ProcessNoLossChangeBitRate) {
// Bitrate and frame rate profile.
RateProfile rate_profile;
SetRateProfilePars(&rate_profile, 0, 200, 30, 0);
SetRateProfilePars(&rate_profile, 1, 800, 30, 100);
SetRateProfilePars(&rate_profile, 2, 500, 30, 200);
rate_profile.frame_index_rate_update[3] = kNbrFramesLong + 1;
rate_profile.num_frames = kNbrFramesLong;
// Codec/network settings.
CodecConfigPars process_settings;
SetCodecParameters(&process_settings, 0.0f, -1, 1, true, true, true, false);
// Metrics for expected quality.
QualityMetrics quality_metrics;
SetQualityMetrics(&quality_metrics, 34.0, 32.0, 0.85, 0.80);
// Metrics for rate control.
RateControlMetrics rc_metrics[3];
SetRateControlMetrics(rc_metrics, 0, 0, 45, 20, 10, 15, 0);
SetRateControlMetrics(rc_metrics, 1, 0, 0, 25, 20, 10, 0);
SetRateControlMetrics(rc_metrics, 2, 0, 0, 25, 15, 10, 0);
ProcessFramesAndVerify(quality_metrics,
rate_profile,
process_settings,
rc_metrics);
}
// Run with no packet loss, with an update (decrease) in frame rate.
// Lower frame rate means higher per-frame-bandwidth, so easier to encode.
// At the bitrate in this test, this means better rate control after the
// update(s) to lower frame rate. So expect less frame drops, and max values
// for the rate control metrics can be lower. One key frame (first frame only).
// Note: quality after update should be higher but we currently compute quality
// metrics avergaed over whole sequence run.
TEST_F(VideoProcessorIntegrationTest, ProcessNoLossChangeFrameRateFrameDrop) {
config_.networking_config.packet_loss_probability = 0;
// Bitrate and frame rate profile.
RateProfile rate_profile;
SetRateProfilePars(&rate_profile, 0, 80, 24, 0);
SetRateProfilePars(&rate_profile, 1, 80, 15, 100);
SetRateProfilePars(&rate_profile, 2, 80, 10, 200);
rate_profile.frame_index_rate_update[3] = kNbrFramesLong + 1;
rate_profile.num_frames = kNbrFramesLong;
// Codec/network settings.
CodecConfigPars process_settings;
SetCodecParameters(&process_settings, 0.0f, -1, 1, true, true, true, false);
// Metrics for expected quality.
QualityMetrics quality_metrics;
SetQualityMetrics(&quality_metrics, 31.0, 23.0, 0.80, 0.65);
quality_metrics.minimum_avg_psnr = 31;
quality_metrics.minimum_min_psnr = 23;
quality_metrics.minimum_avg_ssim = 0.8;
quality_metrics.minimum_min_ssim = 0.65;
// Metrics for rate control.
RateControlMetrics rc_metrics[3];
SetRateControlMetrics(rc_metrics, 0, 40, 20, 75, 15, 60, 0);
SetRateControlMetrics(rc_metrics, 1, 10, 0, 25, 10, 35, 0);
SetRateControlMetrics(rc_metrics, 2, 0, 0, 20, 10, 15, 0);
ProcessFramesAndVerify(quality_metrics,
rate_profile,
process_settings,
rc_metrics);
}
// Run with no packet loss, at low bitrate, then increase rate somewhat.
// Key frame is thrown in every 120 frames. Can expect some frame drops after
// key frame, even at high rate. The internal spatial resizer is on, so expect
// spatial resize down at first key frame, and back up at second key frame.
// Error_concealment is off in this test since there is a memory leak with
// resizing and error concealment.
TEST_F(VideoProcessorIntegrationTest, ProcessNoLossSpatialResizeFrameDrop) {
config_.networking_config.packet_loss_probability = 0;
// Bitrate and frame rate profile.
RateProfile rate_profile;
SetRateProfilePars(&rate_profile, 0, 100, 30, 0);
SetRateProfilePars(&rate_profile, 1, 200, 30, 120);
SetRateProfilePars(&rate_profile, 2, 200, 30, 240);
rate_profile.frame_index_rate_update[3] = kNbrFramesLong + 1;
rate_profile.num_frames = kNbrFramesLong;
// Codec/network settings.
CodecConfigPars process_settings;
SetCodecParameters(&process_settings, 0.0f, 120, 1, false, true, true, true);
// Metrics for expected quality.: lower quality on average from up-sampling
// the down-sampled portion of the run, in case resizer is on.
QualityMetrics quality_metrics;
SetQualityMetrics(&quality_metrics, 29.0, 20.0, 0.75, 0.60);
// Metrics for rate control.
RateControlMetrics rc_metrics[3];
SetRateControlMetrics(rc_metrics, 0, 45, 30, 75, 20, 70, 0);
SetRateControlMetrics(rc_metrics, 1, 20, 35, 30, 20, 15, 1);
SetRateControlMetrics(rc_metrics, 2, 0, 30, 30, 15, 25, 1);
ProcessFramesAndVerify(quality_metrics,
rate_profile,
process_settings,
rc_metrics);
}
// Run with no packet loss, with 3 temporal layers, with a rate update in the
// middle of the sequence. The max values for the frame size mismatch and
// encoding rate mismatch are applied to each layer.
// No dropped frames in this test, and internal spatial resizer is off.
// One key frame (first frame only) in sequence, so no spatial resizing.
TEST_F(VideoProcessorIntegrationTest, ProcessNoLossTemporalLayers) {
config_.networking_config.packet_loss_probability = 0;
// Bitrate and frame rate profile.
RateProfile rate_profile;
SetRateProfilePars(&rate_profile, 0, 200, 30, 0);
SetRateProfilePars(&rate_profile, 1, 400, 30, 150);
rate_profile.frame_index_rate_update[2] = kNbrFramesLong + 1;
rate_profile.num_frames = kNbrFramesLong;
// Codec/network settings.
CodecConfigPars process_settings;
SetCodecParameters(&process_settings, 0.0f, -1, 3, true, true, true, false);
// Metrics for expected quality.
QualityMetrics quality_metrics;
SetQualityMetrics(&quality_metrics, 32.5, 30.0, 0.85, 0.80);
// Metrics for rate control.
RateControlMetrics rc_metrics[2];
SetRateControlMetrics(rc_metrics, 0, 0, 20, 30, 10, 10, 0);
SetRateControlMetrics(rc_metrics, 1, 0, 0, 30, 15, 10, 0);
ProcessFramesAndVerify(quality_metrics,
rate_profile,
process_settings,
rc_metrics);
}
} // namespace webrtc

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "gtest/gtest.h"
#include "gmock/gmock.h"
#include "modules/video_coding/codecs/test/mock/mock_packet_manipulator.h"
#include "modules/video_coding/codecs/test/videoprocessor.h"
#include "modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h"
#include "modules/video_coding/main/interface/video_coding.h"
#include "testsupport/mock/mock_frame_reader.h"
#include "testsupport/mock/mock_frame_writer.h"
#include "testsupport/packet_reader.h"
#include "testsupport/unittest_utils.h"
#include "typedefs.h"
using ::testing::_;
using ::testing::AtLeast;
using ::testing::Return;
namespace webrtc {
namespace test {
// Very basic testing for VideoProcessor. It's mostly tested by running the
// video_quality_measurement program.
class VideoProcessorTest: public testing::Test {
protected:
MockVideoEncoder encoder_mock_;
MockVideoDecoder decoder_mock_;
MockFrameReader frame_reader_mock_;
MockFrameWriter frame_writer_mock_;
MockPacketManipulator packet_manipulator_mock_;
Stats stats_;
TestConfig config_;
VideoCodec codec_settings_;
VideoProcessorTest() {}
virtual ~VideoProcessorTest() {}
void SetUp() {
// Get a codec configuration struct and configure it.
VideoCodingModule::Codec(kVideoCodecVP8, &codec_settings_);
config_.codec_settings = &codec_settings_;
config_.codec_settings->startBitrate = 100;
config_.codec_settings->width = 352;
config_.codec_settings->height = 288;
}
void TearDown() {}
void ExpectInit() {
EXPECT_CALL(encoder_mock_, InitEncode(_, _, _))
.Times(1);
EXPECT_CALL(encoder_mock_, RegisterEncodeCompleteCallback(_))
.Times(AtLeast(1));
EXPECT_CALL(decoder_mock_, InitDecode(_, _))
.Times(1);
EXPECT_CALL(decoder_mock_, RegisterDecodeCompleteCallback(_))
.Times(AtLeast(1));
EXPECT_CALL(frame_reader_mock_, NumberOfFrames())
.WillOnce(Return(1));
EXPECT_CALL(frame_reader_mock_, FrameLength())
.WillOnce(Return(150000));
}
};
TEST_F(VideoProcessorTest, Init) {
ExpectInit();
VideoProcessorImpl video_processor(&encoder_mock_, &decoder_mock_,
&frame_reader_mock_,
&frame_writer_mock_,
&packet_manipulator_mock_, config_,
&stats_);
ASSERT_TRUE(video_processor.Init());
}
TEST_F(VideoProcessorTest, ProcessFrame) {
ExpectInit();
EXPECT_CALL(encoder_mock_, Encode(_, _, _))
.Times(1);
EXPECT_CALL(frame_reader_mock_, ReadFrame(_))
.WillOnce(Return(true));
// Since we don't return any callback from the mock, the decoder will not
// be more than initialized...
VideoProcessorImpl video_processor(&encoder_mock_, &decoder_mock_,
&frame_reader_mock_,
&frame_writer_mock_,
&packet_manipulator_mock_, config_,
&stats_);
ASSERT_TRUE(video_processor.Init());
video_processor.ProcessFrame(0);
}
} // namespace test
} // namespace webrtc

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "benchmark.h"
#include <cassert>
#include <iostream>
#include <sstream>
#include <vector>
#if defined(_WIN32)
#include <windows.h>
#endif
#include "common_video/libyuv/include/webrtc_libyuv.h"
#include "system_wrappers/interface/event_wrapper.h"
#include "modules/video_coding/codecs/test_framework/video_source.h"
#include "testsupport/fileutils.h"
#include "testsupport/metrics/video_metrics.h"
#define SSIM_CALC 0 // by default, don't compute SSIM
using namespace webrtc;
Benchmark::Benchmark()
:
NormalAsyncTest("Benchmark", "Codec benchmark over a range of test cases", 6),
_resultsFileName(webrtc::test::OutputPath() + "benchmark.txt"),
_codecName("Default")
{
}
Benchmark::Benchmark(std::string name, std::string description)
:
NormalAsyncTest(name, description, 6),
_resultsFileName(webrtc::test::OutputPath() + "benchmark.txt"),
_codecName("Default")
{
}
Benchmark::Benchmark(std::string name, std::string description, std::string resultsFileName, std::string codecName)
:
NormalAsyncTest(name, description, 6),
_resultsFileName(resultsFileName),
_codecName(codecName)
{
}
void
Benchmark::Perform()
{
std::vector<const VideoSource*> sources;
std::vector<const VideoSource*>::iterator it;
// Configuration --------------------------
sources.push_back(new const VideoSource(webrtc::test::ProjectRootPath() +
"resources/foreman_cif.yuv", kCIF));
// sources.push_back(new const VideoSource(webrtc::test::ProjectRootPath() +
// "resources/akiyo_cif.yuv", kCIF));
const VideoSize size[] = {kQCIF, kCIF};
const int frameRate[] = {10, 15, 30};
// Specifies the framerates for which to perform a speed test.
const bool speedTestMask[] = {false, false, false};
const int bitRate[] = {50, 100, 200, 300, 400, 500, 600, 1000};
// Determines the number of iterations to perform to arrive at the speed result.
enum { kSpeedTestIterations = 10 };
// ----------------------------------------
const int nFrameRates = sizeof(frameRate)/sizeof(*frameRate);
assert(sizeof(speedTestMask)/sizeof(*speedTestMask) == nFrameRates);
const int nBitrates = sizeof(bitRate)/sizeof(*bitRate);
int testIterations = 10;
webrtc::test::QualityMetricsResult psnr[nBitrates];
webrtc::test::QualityMetricsResult ssim[nBitrates];
double fps[nBitrates];
double totalEncodeTime[nBitrates];
double totalDecodeTime[nBitrates];
_results.open(_resultsFileName.c_str(), std::fstream::out);
_results << GetMagicStr() << std::endl;
_results << _codecName << std::endl;
for (it = sources.begin() ; it < sources.end(); it++)
{
for (int i = 0; i < static_cast<int>(sizeof(size)/sizeof(*size)); i++)
{
for (int j = 0; j < nFrameRates; j++)
{
std::stringstream ss;
std::string strFrameRate;
std::string outFileName;
ss << frameRate[j];
ss >> strFrameRate;
outFileName = (*it)->GetFilePath() + "/" + (*it)->GetName() + "_" +
VideoSource::GetSizeString(size[i]) + "_" + strFrameRate + ".yuv";
_target = new const VideoSource(outFileName, size[i], frameRate[j]);
(*it)->Convert(*_target);
if (VideoSource::FileExists(outFileName.c_str()))
{
_inname = outFileName;
}
else
{
_inname = (*it)->GetFileName();
}
std::cout << (*it)->GetName() << ", " << VideoSource::GetSizeString(size[i])
<< ", " << frameRate[j] << " fps" << std::endl << "Bitrate [kbps]:";
_results << (*it)->GetName() << "," << VideoSource::GetSizeString(size[i])
<< "," << frameRate[j] << " fps" << std::endl << "Bitrate [kbps]";
if (speedTestMask[j])
{
testIterations = kSpeedTestIterations;
}
else
{
testIterations = 1;
}
for (int k = 0; k < nBitrates; k++)
{
_bitRate = (bitRate[k]);
double avgFps = 0.0;
totalEncodeTime[k] = 0;
totalDecodeTime[k] = 0;
for (int l = 0; l < testIterations; l++)
{
PerformNormalTest();
_appendNext = false;
avgFps += _framecnt / (_totalEncodeTime + _totalDecodeTime);
totalEncodeTime[k] += _totalEncodeTime;
totalDecodeTime[k] += _totalDecodeTime;
}
avgFps /= testIterations;
totalEncodeTime[k] /= testIterations;
totalDecodeTime[k] /= testIterations;
double actualBitRate = ActualBitRate(_framecnt) / 1000.0;
std::cout << " " << actualBitRate;
_results << "," << actualBitRate;
webrtc::test::QualityMetricsResult psnr_result;
I420PSNRFromFiles(_inname.c_str(), _outname.c_str(),
_inst.width, _inst.height, &psnr[k]);
if (SSIM_CALC)
{
webrtc::test::QualityMetricsResult ssim_result;
I420SSIMFromFiles(_inname.c_str(), _outname.c_str(),
_inst.width, _inst.height, &ssim[k]);
}
fps[k] = avgFps;
}
std::cout << std::endl << "Y-PSNR [dB]:";
_results << std::endl << "Y-PSNR [dB]";
for (int k = 0; k < nBitrates; k++)
{
std::cout << " " << psnr[k].average;
_results << "," << psnr[k].average;
}
if (SSIM_CALC)
{
std::cout << std::endl << "SSIM: ";
_results << std::endl << "SSIM ";
for (int k = 0; k < nBitrates; k++)
{
std::cout << " " << ssim[k].average;
_results << "," << ssim[k].average;
}
}
std::cout << std::endl << "Encode Time[ms]:";
_results << std::endl << "Encode Time[ms]";
for (int k = 0; k < nBitrates; k++)
{
std::cout << " " << totalEncodeTime[k];
_results << "," << totalEncodeTime[k];
}
std::cout << std::endl << "Decode Time[ms]:";
_results << std::endl << "Decode Time[ms]";
for (int k = 0; k < nBitrates; k++)
{
std::cout << " " << totalDecodeTime[k];
_results << "," << totalDecodeTime[k];
}
if (speedTestMask[j])
{
std::cout << std::endl << "Speed [fps]:";
_results << std::endl << "Speed [fps]";
for (int k = 0; k < nBitrates; k++)
{
std::cout << " " << static_cast<int>(fps[k] + 0.5);
_results << "," << static_cast<int>(fps[k] + 0.5);
}
}
std::cout << std::endl << std::endl;
_results << std::endl << std::endl;
delete _target;
}
}
delete *it;
}
_results.close();
}
void
Benchmark::PerformNormalTest()
{
_encoder = GetNewEncoder();
_decoder = GetNewDecoder();
CodecSettings(_target->GetWidth(), _target->GetHeight(), _target->GetFrameRate(), _bitRate);
Setup();
EventWrapper* waitEvent = EventWrapper::Create();
_inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
_decodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
_encoder->InitEncode(&_inst, 4, 1440);
CodecSpecific_InitBitrate();
_decoder->InitDecode(&_inst,1);
FrameQueue frameQueue;
VideoEncodeCompleteCallback encCallback(_encodedFile, &frameQueue, *this);
VideoDecodeCompleteCallback decCallback(_decodedFile, *this);
_encoder->RegisterEncodeCompleteCallback(&encCallback);
_decoder->RegisterDecodeCompleteCallback(&decCallback);
SetCodecSpecificParameters();
_totalEncodeTime = _totalDecodeTime = 0;
_totalEncodePipeTime = _totalDecodePipeTime = 0;
bool complete = false;
_framecnt = 0;
_encFrameCnt = 0;
_sumEncBytes = 0;
_lengthEncFrame = 0;
while (!complete)
{
complete = Encode();
if (!frameQueue.Empty() || complete)
{
while (!frameQueue.Empty())
{
_frameToDecode = static_cast<FrameQueueTuple *>(frameQueue.PopFrame());
DoPacketLoss();
int ret = Decode();
delete _frameToDecode;
_frameToDecode = NULL;
if (ret < 0)
{
fprintf(stderr,"\n\nError in decoder: %d\n\n", ret);
exit(EXIT_FAILURE);
}
else if (ret == 0)
{
_framecnt++;
}
else
{
fprintf(stderr, "\n\nPositive return value from decode!\n\n");
}
}
}
waitEvent->Wait(5);
}
_inputVideoBuffer.Free();
_encodedVideoBuffer.Free();
_decodedVideoBuffer.Free();
_encoder->Release();
_decoder->Release();
delete waitEvent;
delete _encoder;
delete _decoder;
Teardown();
}
void
Benchmark::CodecSpecific_InitBitrate()
{
if (_bitRate == 0)
{
_encoder->SetRates(600, _inst.maxFramerate);
}
else
{
_encoder->SetRates(_bitRate, _inst.maxFramerate);
}
}

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAWEWORK_BENCHMARK_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAWEWORK_BENCHMARK_H_
#include "normal_async_test.h"
class VideoSource;
class Benchmark : public NormalAsyncTest
{
public:
Benchmark();
virtual void Perform();
protected:
Benchmark(std::string name, std::string description);
Benchmark(std::string name, std::string description, std::string resultsFileName, std::string codecName);
virtual webrtc::VideoEncoder* GetNewEncoder() = 0;
virtual webrtc::VideoDecoder* GetNewDecoder() = 0;
virtual void PerformNormalTest();
virtual void CodecSpecific_InitBitrate();
static const char* GetMagicStr() { return "#!benchmark1.0"; }
const VideoSource* _target;
std::string _resultsFileName;
std::ofstream _results;
std::string _codecName;
};
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAWEWORK_BENCHMARK_H_

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function exportfig(varargin)
%EXPORTFIG Export a figure to Encapsulated Postscript.
% EXPORTFIG(H, FILENAME) writes the figure H to FILENAME. H is
% a figure handle and FILENAME is a string that specifies the
% name of the output file.
%
% EXPORTFIG(...,PARAM1,VAL1,PARAM2,VAL2,...) specifies
% parameters that control various characteristics of the output
% file.
%
% Format Paramter:
% 'Format' one of the strings 'eps','eps2','jpeg','png','preview'
% specifies the output format. Defaults to 'eps'.
% The output format 'preview' does not generate an output
% file but instead creates a new figure window with a
% preview of the exported figure. In this case the
% FILENAME parameter is ignored.
%
% 'Preview' one of the strings 'none', 'tiff'
% specifies a preview for EPS files. Defaults to 'none'.
%
% Size Parameters:
% 'Width' a positive scalar
% specifies the width in the figure's PaperUnits
% 'Height' a positive scalar
% specifies the height in the figure's PaperUnits
%
% Specifying only one dimension sets the other dimension
% so that the exported aspect ratio is the same as the
% figure's current aspect ratio.
% If neither dimension is specified the size defaults to
% the width and height from the figure's PaperPosition.
%
% Rendering Parameters:
% 'Color' one of the strings 'bw', 'gray', 'cmyk'
% 'bw' specifies that lines and text are exported in
% black and all other objects in grayscale
% 'gray' specifies that all objects are exported in grayscale
% 'cmyk' specifies that all objects are exported in color
% using the CMYK color space
% 'Renderer' one of the strings 'painters', 'zbuffer', 'opengl'
% specifies the renderer to use
% 'Resolution' a positive scalar
% specifies the resolution in dots-per-inch.
%
% The default color setting is 'bw'.
%
% Font Parameters:
% 'FontMode' one of the strings 'scaled', 'fixed'
% 'FontSize' a positive scalar
% in 'scaled' mode multiplies with the font size of each
% text object to obtain the exported font size
% in 'fixed' mode specifies the font size of all text
% objects in points
% 'FontEncoding' one of the strings 'latin1', 'adobe'
% specifies the character encoding of the font
%
% If FontMode is 'scaled' but FontSize is not specified then a
% scaling factor is computed from the ratio of the size of the
% exported figure to the size of the actual figure. The minimum
% font size allowed after scaling is 5 points.
% If FontMode is 'fixed' but FontSize is not specified then the
% exported font sizes of all text objects is 7 points.
%
% The default 'FontMode' setting is 'scaled'.
%
% Line Width Parameters:
% 'LineMode' one of the strings 'scaled', 'fixed'
% 'LineWidth' a positive scalar
% the semantics of LineMode and LineWidth are exactly the
% same as FontMode and FontSize, except that they apply
% to line widths instead of font sizes. The minumum line
% width allowed after scaling is 0.5 points.
% If LineMode is 'fixed' but LineWidth is not specified
% then the exported line width of all line objects is 1
% point.
%
% Examples:
% exportfig(gcf,'fig1.eps','height',3);
% Exports the current figure to the file named 'fig1.eps' with
% a height of 3 inches (assuming the figure's PaperUnits is
% inches) and an aspect ratio the same as the figure's aspect
% ratio on screen.
%
% exportfig(gcf, 'fig2.eps', 'FontMode', 'fixed',...
% 'FontSize', 10, 'color', 'cmyk' );
% Exports the current figure to 'fig2.eps' in color with all
% text in 10 point fonts. The size of the exported figure is
% the figure's PaperPostion width and height.
if (nargin < 2)
error('Too few input arguments');
end
% exportfig(H, filename, ...)
H = varargin{1};
if ~ishandle(H) | ~strcmp(get(H,'type'), 'figure')
error('First argument must be a handle to a figure.');
end
filename = varargin{2};
if ~ischar(filename)
error('Second argument must be a string.');
end
paramPairs = varargin(3:end);
% Do some validity checking on param-value pairs
if (rem(length(paramPairs),2) ~= 0)
error(['Invalid input syntax. Optional parameters and values' ...
' must be in pairs.']);
end
format = 'eps';
preview = 'none';
width = -1;
height = -1;
color = 'bw';
fontsize = -1;
fontmode='scaled';
linewidth = -1;
linemode=[];
fontencoding = 'latin1';
renderer = [];
resolution = [];
% Process param-value pairs
args = {};
for k = 1:2:length(paramPairs)
param = lower(paramPairs{k});
if (~ischar(param))
error('Optional parameter names must be strings');
end
value = paramPairs{k+1};
switch (param)
case 'format'
format = value;
if (~strcmp(format,{'eps','eps2','jpeg','png','preview'}))
error(['Format must be ''eps'', ''eps2'', ''jpeg'', ''png'' or' ...
' ''preview''.']);
end
case 'preview'
preview = value;
if (~strcmp(preview,{'none','tiff'}))
error('Preview must be ''none'' or ''tiff''.');
end
case 'width'
width = LocalToNum(value);
if(~LocalIsPositiveScalar(width))
error('Width must be a numeric scalar > 0');
end
case 'height'
height = LocalToNum(value);
if(~LocalIsPositiveScalar(height))
error('Height must be a numeric scalar > 0');
end
case 'color'
color = lower(value);
if (~strcmp(color,{'bw','gray','cmyk'}))
error('Color must be ''bw'', ''gray'' or ''cmyk''.');
end
case 'fontmode'
fontmode = lower(value);
if (~strcmp(fontmode,{'scaled','fixed'}))
error('FontMode must be ''scaled'' or ''fixed''.');
end
case 'fontsize'
fontsize = LocalToNum(value);
if(~LocalIsPositiveScalar(fontsize))
error('FontSize must be a numeric scalar > 0');
end
case 'fontencoding'
fontencoding = lower(value);
if (~strcmp(fontencoding,{'latin1','adobe'}))
error('FontEncoding must be ''latin1'' or ''adobe''.');
end
case 'linemode'
linemode = lower(value);
if (~strcmp(linemode,{'scaled','fixed'}))
error('LineMode must be ''scaled'' or ''fixed''.');
end
case 'linewidth'
linewidth = LocalToNum(value);
if(~LocalIsPositiveScalar(linewidth))
error('LineWidth must be a numeric scalar > 0');
end
case 'renderer'
renderer = lower(value);
if (~strcmp(renderer,{'painters','zbuffer','opengl'}))
error('Renderer must be ''painters'', ''zbuffer'' or ''opengl''.');
end
case 'resolution'
resolution = LocalToNum(value);
if ~(isnumeric(value) & (prod(size(value)) == 1) & (value >= 0));
error('Resolution must be a numeric scalar >= 0');
end
otherwise
error(['Unrecognized option ' param '.']);
end
end
allLines = findall(H, 'type', 'line');
allText = findall(H, 'type', 'text');
allAxes = findall(H, 'type', 'axes');
allImages = findall(H, 'type', 'image');
allLights = findall(H, 'type', 'light');
allPatch = findall(H, 'type', 'patch');
allSurf = findall(H, 'type', 'surface');
allRect = findall(H, 'type', 'rectangle');
allFont = [allText; allAxes];
allColor = [allLines; allText; allAxes; allLights];
allMarker = [allLines; allPatch; allSurf];
allEdge = [allPatch; allSurf];
allCData = [allImages; allPatch; allSurf];
old.objs = {};
old.prop = {};
old.values = {};
% Process format and preview parameter
showPreview = strcmp(format,'preview');
if showPreview
format = 'png';
filename = [tempName '.png'];
end
if strncmp(format,'eps',3) & ~strcmp(preview,'none')
args = {args{:}, ['-' preview]};
end
hadError = 0;
try
% Process size parameters
paperPos = get(H, 'PaperPosition');
old = LocalPushOldData(old, H, 'PaperPosition', paperPos);
figureUnits = get(H, 'Units');
set(H, 'Units', get(H,'PaperUnits'));
figurePos = get(H, 'Position');
aspectRatio = figurePos(3)/figurePos(4);
set(H, 'Units', figureUnits);
if (width == -1) & (height == -1)
width = paperPos(3);
height = paperPos(4);
elseif (width == -1)
width = height * aspectRatio;
elseif (height == -1)
height = width / aspectRatio;
end
set(H, 'PaperPosition', [0 0 width height]);
paperPosMode = get(H, 'PaperPositionMode');
old = LocalPushOldData(old, H, 'PaperPositionMode', paperPosMode);
set(H, 'PaperPositionMode', 'manual');
% Process rendering parameters
switch (color)
case {'bw', 'gray'}
if ~strcmp(color,'bw') & strncmp(format,'eps',3)
format = [format 'c'];
end
args = {args{:}, ['-d' format]};
%compute and set gray colormap
oldcmap = get(H,'Colormap');
newgrays = 0.30*oldcmap(:,1) + 0.59*oldcmap(:,2) + 0.11*oldcmap(:,3);
newcmap = [newgrays newgrays newgrays];
old = LocalPushOldData(old, H, 'Colormap', oldcmap);
set(H, 'Colormap', newcmap);
%compute and set ColorSpec and CData properties
old = LocalUpdateColors(allColor, 'color', old);
old = LocalUpdateColors(allAxes, 'xcolor', old);
old = LocalUpdateColors(allAxes, 'ycolor', old);
old = LocalUpdateColors(allAxes, 'zcolor', old);
old = LocalUpdateColors(allMarker, 'MarkerEdgeColor', old);
old = LocalUpdateColors(allMarker, 'MarkerFaceColor', old);
old = LocalUpdateColors(allEdge, 'EdgeColor', old);
old = LocalUpdateColors(allEdge, 'FaceColor', old);
old = LocalUpdateColors(allCData, 'CData', old);
case 'cmyk'
if strncmp(format,'eps',3)
format = [format 'c'];
args = {args{:}, ['-d' format], '-cmyk'};
else
args = {args{:}, ['-d' format]};
end
otherwise
error('Invalid Color parameter');
end
if (~isempty(renderer))
args = {args{:}, ['-' renderer]};
end
if (~isempty(resolution)) | ~strncmp(format,'eps',3)
if isempty(resolution)
resolution = 0;
end
args = {args{:}, ['-r' int2str(resolution)]};
end
% Process font parameters
if (~isempty(fontmode))
oldfonts = LocalGetAsCell(allFont,'FontSize');
switch (fontmode)
case 'fixed'
oldfontunits = LocalGetAsCell(allFont,'FontUnits');
old = LocalPushOldData(old, allFont, {'FontUnits'}, oldfontunits);
set(allFont,'FontUnits','points');
if (fontsize == -1)
set(allFont,'FontSize',7);
else
set(allFont,'FontSize',fontsize);
end
case 'scaled'
if (fontsize == -1)
wscale = width/figurePos(3);
hscale = height/figurePos(4);
scale = min(wscale, hscale);
else
scale = fontsize;
end
newfonts = LocalScale(oldfonts,scale,5);
set(allFont,{'FontSize'},newfonts);
otherwise
error('Invalid FontMode parameter');
end
% make sure we push the size after the units
old = LocalPushOldData(old, allFont, {'FontSize'}, oldfonts);
end
if strcmp(fontencoding,'adobe') & strncmp(format,'eps',3)
args = {args{:}, '-adobecset'};
end
% Process linewidth parameters
if (~isempty(linemode))
oldlines = LocalGetAsCell(allMarker,'LineWidth');
old = LocalPushOldData(old, allMarker, {'LineWidth'}, oldlines);
switch (linemode)
case 'fixed'
if (linewidth == -1)
set(allMarker,'LineWidth',1);
else
set(allMarker,'LineWidth',linewidth);
end
case 'scaled'
if (linewidth == -1)
wscale = width/figurePos(3);
hscale = height/figurePos(4);
scale = min(wscale, hscale);
else
scale = linewidth;
end
newlines = LocalScale(oldlines, scale, 0.5);
set(allMarker,{'LineWidth'},newlines);
otherwise
error('Invalid LineMode parameter');
end
end
% Export
print(H, filename, args{:});
catch
hadError = 1;
end
% Restore figure settings
for n=1:length(old.objs)
set(old.objs{n}, old.prop{n}, old.values{n});
end
if hadError
error(deblank(lasterr));
end
% Show preview if requested
if showPreview
X = imread(filename,'png');
delete(filename);
f = figure( 'Name', 'Preview', ...
'Menubar', 'none', ...
'NumberTitle', 'off', ...
'Visible', 'off');
image(X);
axis image;
ax = findobj(f, 'type', 'axes');
set(ax, 'Units', get(H,'PaperUnits'), ...
'Position', [0 0 width height], ...
'Visible', 'off');
set(ax, 'Units', 'pixels');
axesPos = get(ax,'Position');
figPos = get(f,'Position');
rootSize = get(0,'ScreenSize');
figPos(3:4) = axesPos(3:4);
if figPos(1) + figPos(3) > rootSize(3)
figPos(1) = rootSize(3) - figPos(3) - 50;
end
if figPos(2) + figPos(4) > rootSize(4)
figPos(2) = rootSize(4) - figPos(4) - 50;
end
set(f, 'Position',figPos, ...
'Visible', 'on');
end
%
% Local Functions
%
function outData = LocalPushOldData(inData, objs, prop, values)
outData.objs = {inData.objs{:}, objs};
outData.prop = {inData.prop{:}, prop};
outData.values = {inData.values{:}, values};
function cellArray = LocalGetAsCell(fig,prop);
cellArray = get(fig,prop);
if (~isempty(cellArray)) & (~iscell(cellArray))
cellArray = {cellArray};
end
function newArray = LocalScale(inArray, scale, minValue)
n = length(inArray);
newArray = cell(n,1);
for k=1:n
newArray{k} = max(minValue,scale*inArray{k}(1));
end
function newArray = LocalMapToGray(inArray);
n = length(inArray);
newArray = cell(n,1);
for k=1:n
color = inArray{k};
if (~isempty(color))
if ischar(color)
switch color(1)
case 'y'
color = [1 1 0];
case 'm'
color = [1 0 1];
case 'c'
color = [0 1 1];
case 'r'
color = [1 0 0];
case 'g'
color = [0 1 0];
case 'b'
color = [0 0 1];
case 'w'
color = [1 1 1];
case 'k'
color = [0 0 0];
otherwise
newArray{k} = color;
end
end
if ~ischar(color)
color = 0.30*color(1) + 0.59*color(2) + 0.11*color(3);
end
end
if isempty(color) | ischar(color)
newArray{k} = color;
else
newArray{k} = [color color color];
end
end
function newArray = LocalMapCData(inArray);
n = length(inArray);
newArray = cell(n,1);
for k=1:n
color = inArray{k};
if (ndims(color) == 3) & isa(color,'double')
gray = 0.30*color(:,:,1) + 0.59*color(:,:,2) + 0.11*color(:,:,3);
color(:,:,1) = gray;
color(:,:,2) = gray;
color(:,:,3) = gray;
end
newArray{k} = color;
end
function outData = LocalUpdateColors(inArray, prop, inData)
value = LocalGetAsCell(inArray,prop);
outData.objs = {inData.objs{:}, inArray};
outData.prop = {inData.prop{:}, {prop}};
outData.values = {inData.values{:}, value};
if (~isempty(value))
if strcmp(prop,'CData')
value = LocalMapCData(value);
else
value = LocalMapToGray(value);
end
set(inArray,{prop},value);
end
function bool = LocalIsPositiveScalar(value)
bool = isnumeric(value) & ...
prod(size(value)) == 1 & ...
value > 0;
function value = LocalToNum(value)
if ischar(value)
value = str2num(value);
end

View File

@@ -0,0 +1,600 @@
/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "normal_async_test.h"
#include <assert.h>
#include <string.h>
#include <queue>
#include <sstream>
#include <vector>
#include "gtest/gtest.h"
#include "tick_util.h"
#include "testsupport/fileutils.h"
#include "typedefs.h"
using namespace webrtc;
NormalAsyncTest::NormalAsyncTest()
:
NormalTest("Async Normal Test 1", "A test of normal execution of the codec",
_testNo),
_decodeCompleteTime(0),
_encodeCompleteTime(0),
_encFrameCnt(0),
_decFrameCnt(0),
_requestKeyFrame(false),
_testNo(1),
_appendNext(false),
_missingFrames(false),
_rttFrames(0),
_hasReceivedSLI(false),
_hasReceivedRPSI(false),
_hasReceivedPLI(false),
_waitForKey(false)
{
}
NormalAsyncTest::NormalAsyncTest(WebRtc_UWord32 bitRate)
:
NormalTest("Async Normal Test 1", "A test of normal execution of the codec",
bitRate, _testNo),
_decodeCompleteTime(0),
_encodeCompleteTime(0),
_encFrameCnt(0),
_decFrameCnt(0),
_requestKeyFrame(false),
_testNo(1),
_appendNext(false),
_missingFrames(false),
_rttFrames(0),
_hasReceivedSLI(false),
_hasReceivedRPSI(false),
_hasReceivedPLI(false),
_waitForKey(false)
{
}
NormalAsyncTest::NormalAsyncTest(std::string name, std::string description,
unsigned int testNo)
:
NormalTest(name, description, _testNo),
_decodeCompleteTime(0),
_encodeCompleteTime(0),
_encFrameCnt(0),
_decFrameCnt(0),
_requestKeyFrame(false),
_testNo(testNo),
_lengthEncFrame(0),
_appendNext(false),
_missingFrames(false),
_rttFrames(0),
_hasReceivedSLI(false),
_hasReceivedRPSI(false),
_hasReceivedPLI(false),
_waitForKey(false)
{
}
NormalAsyncTest::NormalAsyncTest(std::string name, std::string description,
WebRtc_UWord32 bitRate, unsigned int testNo)
:
NormalTest(name, description, bitRate, _testNo),
_decodeCompleteTime(0),
_encodeCompleteTime(0),
_encFrameCnt(0),
_decFrameCnt(0),
_requestKeyFrame(false),
_testNo(testNo),
_lengthEncFrame(0),
_appendNext(false),
_missingFrames(false),
_rttFrames(0),
_hasReceivedSLI(false),
_hasReceivedRPSI(false),
_hasReceivedPLI(false),
_waitForKey(false)
{
}
NormalAsyncTest::NormalAsyncTest(std::string name, std::string description,
WebRtc_UWord32 bitRate, unsigned int testNo,
unsigned int rttFrames)
:
NormalTest(name, description, bitRate, _testNo),
_decodeCompleteTime(0),
_encodeCompleteTime(0),
_encFrameCnt(0),
_decFrameCnt(0),
_requestKeyFrame(false),
_testNo(testNo),
_lengthEncFrame(0),
_appendNext(false),
_missingFrames(false),
_rttFrames(rttFrames),
_hasReceivedSLI(false),
_hasReceivedRPSI(false),
_hasReceivedPLI(false),
_waitForKey(false)
{
}
void
NormalAsyncTest::Setup()
{
CodecTest::Setup();
std::stringstream ss;
std::string strTestNo;
ss << _testNo;
ss >> strTestNo;
// Check if settings exist. Otherwise use defaults.
if (_outname == "")
{
_outname = webrtc::test::OutputPath() + "out_normaltest" + strTestNo +
".yuv";
}
if (_encodedName == "")
{
_encodedName = webrtc::test::OutputPath() + "encoded_normaltest" +
strTestNo + ".yuv";
}
if ((_sourceFile = fopen(_inname.c_str(), "rb")) == NULL)
{
printf("Cannot read file %s.\n", _inname.c_str());
exit(1);
}
if ((_encodedFile = fopen(_encodedName.c_str(), "wb")) == NULL)
{
printf("Cannot write encoded file.\n");
exit(1);
}
char mode[3] = "wb";
if (_appendNext)
{
strncpy(mode, "ab", 3);
}
if ((_decodedFile = fopen(_outname.c_str(), mode)) == NULL)
{
printf("Cannot write file %s.\n", _outname.c_str());
exit(1);
}
_appendNext = true;
}
void
NormalAsyncTest::Teardown()
{
CodecTest::Teardown();
fclose(_sourceFile);
fclose(_encodedFile);
fclose(_decodedFile);
}
FrameQueueTuple::~FrameQueueTuple()
{
if (_codecSpecificInfo != NULL)
{
delete _codecSpecificInfo;
}
if (_frame != NULL)
{
delete _frame;
}
}
void FrameQueue::PushFrame(VideoFrame *frame,
webrtc::CodecSpecificInfo* codecSpecificInfo)
{
WriteLockScoped cs(_queueRWLock);
_frameBufferQueue.push(new FrameQueueTuple(frame, codecSpecificInfo));
}
FrameQueueTuple* FrameQueue::PopFrame()
{
WriteLockScoped cs(_queueRWLock);
if (_frameBufferQueue.empty())
{
return NULL;
}
FrameQueueTuple* tuple = _frameBufferQueue.front();
_frameBufferQueue.pop();
return tuple;
}
bool FrameQueue::Empty()
{
ReadLockScoped cs(_queueRWLock);
return _frameBufferQueue.empty();
}
WebRtc_UWord32 VideoEncodeCompleteCallback::EncodedBytes()
{
return _encodedBytes;
}
WebRtc_Word32
VideoEncodeCompleteCallback::Encoded(EncodedImage& encodedImage,
const webrtc::CodecSpecificInfo* codecSpecificInfo,
const webrtc::RTPFragmentationHeader*
fragmentation)
{
_test.Encoded(encodedImage);
VideoFrame *newBuffer = new VideoFrame();
//newBuffer->VerifyAndAllocate(encodedImage._length);
newBuffer->VerifyAndAllocate(encodedImage._size);
_encodedBytes += encodedImage._length;
// If _frameQueue would have been a fixed sized buffer we could have asked
// it for an empty frame and then just do:
// emptyFrame->SwapBuffers(encodedBuffer);
// This is how it should be done in Video Engine to save in on memcpys
webrtc::CodecSpecificInfo* codecSpecificInfoCopy =
_test.CopyCodecSpecificInfo(codecSpecificInfo);
_test.CopyEncodedImage(*newBuffer, encodedImage, codecSpecificInfoCopy);
if (_encodedFile != NULL)
{
if (fwrite(newBuffer->Buffer(), 1, newBuffer->Length(),
_encodedFile) != newBuffer->Length()) {
return -1;
}
}
_frameQueue->PushFrame(newBuffer, codecSpecificInfoCopy);
return 0;
}
WebRtc_UWord32 VideoDecodeCompleteCallback::DecodedBytes()
{
return _decodedBytes;
}
WebRtc_Word32
VideoDecodeCompleteCallback::Decoded(VideoFrame& image)
{
_test.Decoded(image);
_decodedBytes += image.Length();
if (_decodedFile != NULL)
{
if (fwrite(image.Buffer(), 1, image.Length(),
_decodedFile) != image.Length()) {
return -1;
}
}
return 0;
}
WebRtc_Word32
VideoDecodeCompleteCallback::ReceivedDecodedReferenceFrame(
const WebRtc_UWord64 pictureId)
{
return _test.ReceivedDecodedReferenceFrame(pictureId);
}
WebRtc_Word32
VideoDecodeCompleteCallback::ReceivedDecodedFrame(
const WebRtc_UWord64 pictureId)
{
return _test.ReceivedDecodedFrame(pictureId);
}
void
NormalAsyncTest::Encoded(const EncodedImage& encodedImage)
{
_encodeCompleteTime = tGetTime();
_encFrameCnt++;
_totalEncodePipeTime += _encodeCompleteTime -
_encodeTimes[encodedImage._timeStamp];
}
void
NormalAsyncTest::Decoded(const VideoFrame& decodedImage)
{
_decodeCompleteTime = tGetTime();
_decFrameCnt++;
_totalDecodePipeTime += _decodeCompleteTime -
_decodeTimes[decodedImage.TimeStamp()];
_decodedWidth = decodedImage.Width();
_decodedHeight = decodedImage.Height();
}
void
NormalAsyncTest::Perform()
{
_inname = webrtc::test::ProjectRootPath() + "resources/foreman_cif.yuv";
CodecSettings(352, 288, 30, _bitRate);
Setup();
_inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
_decodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
if(_encoder->InitEncode(&_inst, 1, 1440) < 0)
{
exit(EXIT_FAILURE);
}
_decoder->InitDecode(&_inst, 1);
FrameQueue frameQueue;
VideoEncodeCompleteCallback encCallback(_encodedFile, &frameQueue, *this);
VideoDecodeCompleteCallback decCallback(_decodedFile, *this);
_encoder->RegisterEncodeCompleteCallback(&encCallback);
_decoder->RegisterDecodeCompleteCallback(&decCallback);
if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
{
exit(EXIT_FAILURE);
}
_totalEncodeTime = _totalDecodeTime = 0;
_totalEncodePipeTime = _totalDecodePipeTime = 0;
bool complete = false;
_framecnt = 0;
_encFrameCnt = 0;
_decFrameCnt = 0;
_sumEncBytes = 0;
_lengthEncFrame = 0;
double starttime = tGetTime();
while (!complete)
{
CodecSpecific_InitBitrate();
complete = Encode();
if (!frameQueue.Empty() || complete)
{
while (!frameQueue.Empty())
{
_frameToDecode =
static_cast<FrameQueueTuple *>(frameQueue.PopFrame());
int lost = DoPacketLoss();
if (lost == 2)
{
// Lost the whole frame, continue
_missingFrames = true;
delete _frameToDecode;
_frameToDecode = NULL;
continue;
}
int ret = Decode(lost);
delete _frameToDecode;
_frameToDecode = NULL;
if (ret < 0)
{
fprintf(stderr,"\n\nError in decoder: %d\n\n", ret);
exit(EXIT_FAILURE);
}
else if (ret == 0)
{
_framecnt++;
}
else
{
fprintf(stderr,
"\n\nPositive return value from decode!\n\n");
}
}
}
}
double endtime = tGetTime();
double totalExecutionTime = endtime - starttime;
printf("Total execution time: %.1f s\n", totalExecutionTime);
_sumEncBytes = encCallback.EncodedBytes();
double actualBitRate = ActualBitRate(_encFrameCnt) / 1000.0;
double avgEncTime = _totalEncodeTime / _encFrameCnt;
double avgDecTime = _totalDecodeTime / _decFrameCnt;
printf("Actual bitrate: %f kbps\n", actualBitRate);
printf("Average encode time: %.1f ms\n", 1000 * avgEncTime);
printf("Average decode time: %.1f ms\n", 1000 * avgDecTime);
printf("Average encode pipeline time: %.1f ms\n",
1000 * _totalEncodePipeTime / _encFrameCnt);
printf("Average decode pipeline time: %.1f ms\n",
1000 * _totalDecodePipeTime / _decFrameCnt);
printf("Number of encoded frames: %u\n", _encFrameCnt);
printf("Number of decoded frames: %u\n", _decFrameCnt);
(*_log) << "Actual bitrate: " << actualBitRate << " kbps\tTarget: " <<
_bitRate << " kbps" << std::endl;
(*_log) << "Average encode time: " << avgEncTime << " s" << std::endl;
(*_log) << "Average decode time: " << avgDecTime << " s" << std::endl;
_encoder->Release();
_decoder->Release();
Teardown();
}
bool
NormalAsyncTest::Encode()
{
_lengthEncFrame = 0;
EXPECT_GT(fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile), 0u);
_inputVideoBuffer.CopyFrame(_lengthSourceFrame, _sourceBuffer);
_inputVideoBuffer.SetTimeStamp((unsigned int)
(_encFrameCnt * 9e4 / _inst.maxFramerate));
_inputVideoBuffer.SetWidth(_inst.width);
_inputVideoBuffer.SetHeight(_inst.height);
if (feof(_sourceFile) != 0)
{
return true;
}
_encodeCompleteTime = 0;
_encodeTimes[_inputVideoBuffer.TimeStamp()] = tGetTime();
std::vector<VideoFrameType> frame_types(1, kDeltaFrame);
// check SLI queue
_hasReceivedSLI = false;
while (!_signalSLI.empty() && _signalSLI.front().delay == 0)
{
// SLI message has arrived at sender side
_hasReceivedSLI = true;
_pictureIdSLI = _signalSLI.front().id;
_signalSLI.pop_front();
}
// decrement SLI queue times
for (std::list<fbSignal>::iterator it = _signalSLI.begin();
it !=_signalSLI.end(); it++)
{
(*it).delay--;
}
// check PLI queue
_hasReceivedPLI = false;
while (!_signalPLI.empty() && _signalPLI.front().delay == 0)
{
// PLI message has arrived at sender side
_hasReceivedPLI = true;
_signalPLI.pop_front();
}
// decrement PLI queue times
for (std::list<fbSignal>::iterator it = _signalPLI.begin();
it != _signalPLI.end(); it++)
{
(*it).delay--;
}
if (_hasReceivedPLI)
{
// respond to PLI by encoding a key frame
frame_types[0] = kKeyFrame;
_hasReceivedPLI = false;
_hasReceivedSLI = false; // don't trigger both at once
}
webrtc::CodecSpecificInfo* codecSpecificInfo = CreateEncoderSpecificInfo();
int ret = _encoder->Encode(_inputVideoBuffer,
codecSpecificInfo, &frame_types);
EXPECT_EQ(ret, WEBRTC_VIDEO_CODEC_OK);
if (codecSpecificInfo != NULL)
{
delete codecSpecificInfo;
codecSpecificInfo = NULL;
}
if (_encodeCompleteTime > 0)
{
_totalEncodeTime += _encodeCompleteTime -
_encodeTimes[_inputVideoBuffer.TimeStamp()];
}
else
{
_totalEncodeTime += tGetTime() -
_encodeTimes[_inputVideoBuffer.TimeStamp()];
}
assert(ret >= 0);
return false;
}
int
NormalAsyncTest::Decode(int lossValue)
{
_sumEncBytes += _frameToDecode->_frame->Length();
EncodedImage encodedImage;
VideoEncodedBufferToEncodedImage(*(_frameToDecode->_frame), encodedImage);
encodedImage._completeFrame = !lossValue;
_decodeCompleteTime = 0;
_decodeTimes[encodedImage._timeStamp] = tGetTime();
int ret = WEBRTC_VIDEO_CODEC_OK;
if (!_waitForKey || encodedImage._frameType == kKeyFrame)
{
_waitForKey = false;
ret = _decoder->Decode(encodedImage, _missingFrames, NULL,
_frameToDecode->_codecSpecificInfo);
if (ret >= 0)
{
_missingFrames = false;
}
}
// check for SLI
if (ret == WEBRTC_VIDEO_CODEC_REQUEST_SLI)
{
// add an SLI feedback to the feedback "queue"
// to be delivered to encoder with _rttFrames delay
_signalSLI.push_back(fbSignal(_rttFrames,
static_cast<WebRtc_UWord8>((_lastDecPictureId) & 0x3f))); // 6 lsb
ret = WEBRTC_VIDEO_CODEC_OK;
}
else if (ret == WEBRTC_VIDEO_CODEC_ERR_REQUEST_SLI)
{
// add an SLI feedback to the feedback "queue"
// to be delivered to encoder with _rttFrames delay
_signalSLI.push_back(fbSignal(_rttFrames,
static_cast<WebRtc_UWord8>((_lastDecPictureId + 1) & 0x3f)));//6 lsb
ret = WEBRTC_VIDEO_CODEC_OK;
}
else if (ret == WEBRTC_VIDEO_CODEC_ERROR)
{
// wait for new key frame
// add an PLI feedback to the feedback "queue"
// to be delivered to encoder with _rttFrames delay
_signalPLI.push_back(fbSignal(_rttFrames, 0 /* picId not used*/));
_waitForKey = true;
ret = WEBRTC_VIDEO_CODEC_OK;
}
if (_decodeCompleteTime > 0)
{
_totalDecodeTime += _decodeCompleteTime -
_decodeTimes[encodedImage._timeStamp];
}
else
{
_totalDecodeTime += tGetTime() - _decodeTimes[encodedImage._timeStamp];
}
return ret;
}
webrtc::CodecSpecificInfo*
NormalAsyncTest::CopyCodecSpecificInfo(
const webrtc::CodecSpecificInfo* codecSpecificInfo) const
{
webrtc::CodecSpecificInfo* info = new webrtc::CodecSpecificInfo;
*info = *codecSpecificInfo;
return info;
}
void NormalAsyncTest::CodecSpecific_InitBitrate()
{
if (_bitRate == 0)
{
_encoder->SetRates(600, _inst.maxFramerate);
}
else
{
_encoder->SetRates(_bitRate, _inst.maxFramerate);
}
}
void NormalAsyncTest::CopyEncodedImage(VideoFrame& dest,
EncodedImage& src,
void* /*codecSpecificInfo*/) const
{
dest.CopyFrame(src._length, src._buffer);
//dest.SetFrameType(src._frameType);
dest.SetWidth((WebRtc_UWord16)src._encodedWidth);
dest.SetHeight((WebRtc_UWord16)src._encodedHeight);
dest.SetTimeStamp(src._timeStamp);
}
WebRtc_Word32 NormalAsyncTest::ReceivedDecodedReferenceFrame(
const WebRtc_UWord64 pictureId) {
_lastDecRefPictureId = pictureId;
return 0;
}
WebRtc_Word32 NormalAsyncTest::ReceivedDecodedFrame(
const WebRtc_UWord64 pictureId) {
_lastDecPictureId = pictureId;
return 0;
}
double
NormalAsyncTest::tGetTime()
{// return time in sec
return ((double) (TickTime::MillisecondTimestamp())/1000);
}

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_NORMAL_ASYNC_TEST_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_NORMAL_ASYNC_TEST_H_
#include "common_types.h"
#include "normal_test.h"
#include "rw_lock_wrapper.h"
#include <list>
#include <map>
#include <queue>
class FrameQueueTuple
{
public:
FrameQueueTuple(webrtc::VideoFrame *frame,
const webrtc::CodecSpecificInfo* codecSpecificInfo = NULL)
:
_frame(frame),
_codecSpecificInfo(codecSpecificInfo)
{};
~FrameQueueTuple();
webrtc::VideoFrame* _frame;
const webrtc::CodecSpecificInfo* _codecSpecificInfo;
};
class FrameQueue
{
public:
FrameQueue()
:
_queueRWLock(*webrtc::RWLockWrapper::CreateRWLock())
{
}
~FrameQueue()
{
delete &_queueRWLock;
}
void PushFrame(webrtc::VideoFrame *frame,
webrtc::CodecSpecificInfo* codecSpecificInfo = NULL);
FrameQueueTuple* PopFrame();
bool Empty();
private:
webrtc::RWLockWrapper& _queueRWLock;
std::queue<FrameQueueTuple *> _frameBufferQueue;
};
// feedback signal to encoder
struct fbSignal
{
fbSignal(int d, WebRtc_UWord8 pid) : delay(d), id(pid) {};
int delay;
WebRtc_UWord8 id;
};
class NormalAsyncTest : public NormalTest
{
public:
NormalAsyncTest();
NormalAsyncTest(WebRtc_UWord32 bitRate);
NormalAsyncTest(std::string name, std::string description,
unsigned int testNo);
NormalAsyncTest(std::string name, std::string description,
WebRtc_UWord32 bitRate, unsigned int testNo);
NormalAsyncTest(std::string name, std::string description,
WebRtc_UWord32 bitRate, unsigned int testNo,
unsigned int rttFrames);
virtual ~NormalAsyncTest() {};
virtual void Perform();
virtual void Encoded(const webrtc::EncodedImage& encodedImage);
virtual void Decoded(const webrtc::VideoFrame& decodedImage);
virtual webrtc::CodecSpecificInfo*
CopyCodecSpecificInfo(
const webrtc::CodecSpecificInfo* codecSpecificInfo) const;
virtual void CopyEncodedImage(webrtc::VideoFrame& dest,
webrtc::EncodedImage& src,
void* /*codecSpecificInfo*/) const;
virtual webrtc::CodecSpecificInfo* CreateEncoderSpecificInfo() const
{
return NULL;
};
virtual WebRtc_Word32 ReceivedDecodedReferenceFrame(
const WebRtc_UWord64 pictureId);
virtual WebRtc_Word32 ReceivedDecodedFrame(const WebRtc_UWord64 pictureId);
protected:
virtual void Setup();
virtual void Teardown();
virtual bool Encode();
virtual int Decode(int lossValue = 0);
virtual void CodecSpecific_InitBitrate();
virtual int SetCodecSpecificParameters() {return 0;};
double tGetTime();// return time in sec
FILE* _sourceFile;
FILE* _decodedFile;
WebRtc_UWord32 _decodedWidth;
WebRtc_UWord32 _decodedHeight;
double _totalEncodeTime;
double _totalDecodeTime;
double _decodeCompleteTime;
double _encodeCompleteTime;
double _totalEncodePipeTime;
double _totalDecodePipeTime;
int _framecnt;
int _encFrameCnt;
int _decFrameCnt;
bool _requestKeyFrame;
unsigned int _testNo;
unsigned int _lengthEncFrame;
FrameQueueTuple* _frameToDecode;
bool _appendNext;
std::map<WebRtc_UWord32, double> _encodeTimes;
std::map<WebRtc_UWord32, double> _decodeTimes;
bool _missingFrames;
std::list<fbSignal> _signalSLI;
int _rttFrames;
mutable bool _hasReceivedSLI;
mutable bool _hasReceivedRPSI;
WebRtc_UWord8 _pictureIdSLI;
WebRtc_UWord16 _pictureIdRPSI;
WebRtc_UWord64 _lastDecRefPictureId;
WebRtc_UWord64 _lastDecPictureId;
std::list<fbSignal> _signalPLI;
bool _hasReceivedPLI;
bool _waitForKey;
};
class VideoEncodeCompleteCallback : public webrtc::EncodedImageCallback
{
public:
VideoEncodeCompleteCallback(FILE* encodedFile, FrameQueue *frameQueue,
NormalAsyncTest& test)
:
_encodedFile(encodedFile),
_frameQueue(frameQueue),
_test(test),
_encodedBytes(0)
{}
WebRtc_Word32
Encoded(webrtc::EncodedImage& encodedImage,
const webrtc::CodecSpecificInfo* codecSpecificInfo = NULL,
const webrtc::RTPFragmentationHeader* fragmentation = NULL);
WebRtc_UWord32 EncodedBytes();
private:
FILE* _encodedFile;
FrameQueue* _frameQueue;
NormalAsyncTest& _test;
WebRtc_UWord32 _encodedBytes;
};
class VideoDecodeCompleteCallback : public webrtc::DecodedImageCallback
{
public:
VideoDecodeCompleteCallback(FILE* decodedFile, NormalAsyncTest& test)
:
_decodedFile(decodedFile),
_test(test),
_decodedBytes(0)
{}
virtual WebRtc_Word32 Decoded(webrtc::VideoFrame& decodedImage);
virtual WebRtc_Word32
ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId);
virtual WebRtc_Word32 ReceivedDecodedFrame(const WebRtc_UWord64 pictureId);
WebRtc_UWord32 DecodedBytes();
private:
FILE* _decodedFile;
NormalAsyncTest& _test;
WebRtc_UWord32 _decodedBytes;
};
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_NORMAL_ASYNC_TEST_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "normal_test.h"
#include <time.h>
#include <sstream>
#include <string.h>
#include "gtest/gtest.h"
#include "testsupport/fileutils.h"
NormalTest::NormalTest()
:
CodecTest("Normal Test 1", "A test of normal execution of the codec"),
_testNo(1),
_lengthEncFrame(0),
_appendNext(false)
{
}
NormalTest::NormalTest(std::string name, std::string description,
unsigned int testNo)
:
CodecTest(name, description),
_requestKeyFrame(false),
_testNo(testNo),
_lengthEncFrame(0),
_appendNext(false)
{
}
NormalTest::NormalTest(std::string name, std::string description,
WebRtc_UWord32 bitRate, unsigned int testNo)
:
CodecTest(name, description, bitRate),
_requestKeyFrame(false),
_testNo(testNo),
_lengthEncFrame(0),
_appendNext(false)
{
}
void
NormalTest::Setup()
{
CodecTest::Setup();
std::stringstream ss;
std::string strTestNo;
ss << _testNo;
ss >> strTestNo;
// Check if settings exist. Otherwise use defaults.
if (_outname == "")
{
_outname = webrtc::test::OutputPath() + "out_normaltest" + strTestNo +
".yuv";
}
if (_encodedName == "")
{
_encodedName = webrtc::test::OutputPath() + "encoded_normaltest" +
strTestNo + ".yuv";
}
if ((_sourceFile = fopen(_inname.c_str(), "rb")) == NULL)
{
printf("Cannot read file %s.\n", _inname.c_str());
exit(1);
}
if ((_encodedFile = fopen(_encodedName.c_str(), "wb")) == NULL)
{
printf("Cannot write encoded file.\n");
exit(1);
}
char mode[3] = "wb";
if (_appendNext)
{
strncpy(mode, "ab", 3);
}
if ((_decodedFile = fopen(_outname.c_str(), mode)) == NULL)
{
printf("Cannot write file %s.\n", _outname.c_str());
exit(1);
}
_appendNext = true;
}
void
NormalTest::Teardown()
{
CodecTest::Teardown();
fclose(_sourceFile);
fclose(_decodedFile);
}
void
NormalTest::Perform()
{
_inname = webrtc::test::ProjectRootPath() + "resources/foreman_cif.yuv";
CodecSettings(352, 288, 30, _bitRate);
Setup();
_inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
_decodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
_encodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
_encoder->InitEncode(&_inst, 1, 1460);
CodecSpecific_InitBitrate();
_decoder->InitDecode(&_inst,1);
_totalEncodeTime = _totalDecodeTime = 0;
_framecnt = 0;
_sumEncBytes = 0;
_lengthEncFrame = 0;
int decodeLength = 0;
while (!Encode())
{
DoPacketLoss();
_encodedVideoBuffer.SetLength(_encodedVideoBuffer.Length());
if (fwrite(_encodedVideoBuffer.Buffer(), 1,
_encodedVideoBuffer.Length(),
_encodedFile) != _encodedVideoBuffer.Length()) {
return;
}
decodeLength = Decode();
if (decodeLength < 0)
{
fprintf(stderr,"\n\nError in decoder: %d\n\n", decodeLength);
exit(EXIT_FAILURE);
}
if (fwrite(_decodedVideoBuffer.Buffer(), 1, decodeLength,
_decodedFile) != static_cast<unsigned int>(decodeLength)) {
return;
}
CodecSpecific_InitBitrate();
_framecnt++;
}
// Ensure we empty the decoding queue.
while (decodeLength > 0)
{
decodeLength = Decode();
if (decodeLength < 0)
{
fprintf(stderr,"\n\nError in decoder: %d\n\n", decodeLength);
exit(EXIT_FAILURE);
}
if (fwrite(_decodedVideoBuffer.Buffer(), 1, decodeLength,
_decodedFile) != static_cast<unsigned int>(decodeLength)) {
return;
}
}
double actualBitRate = ActualBitRate(_framecnt) / 1000.0;
double avgEncTime = _totalEncodeTime / _framecnt;
double avgDecTime = _totalDecodeTime / _framecnt;
printf("Actual bitrate: %f kbps\n", actualBitRate);
printf("Average encode time: %f s\n", avgEncTime);
printf("Average decode time: %f s\n", avgDecTime);
(*_log) << "Actual bitrate: " << actualBitRate << " kbps\tTarget: " << _bitRate << " kbps" << std::endl;
(*_log) << "Average encode time: " << avgEncTime << " s" << std::endl;
(*_log) << "Average decode time: " << avgDecTime << " s" << std::endl;
_inputVideoBuffer.Free();
_encoder->Release();
_decoder->Release();
Teardown();
}
bool
NormalTest::Encode()
{
_lengthEncFrame = 0;
EXPECT_GT(fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile), 0u);
if (feof(_sourceFile) != 0)
{
return true;
}
_inputVideoBuffer.CopyFrame(_lengthSourceFrame, _sourceBuffer);
_inputVideoBuffer.SetTimeStamp(_framecnt);
// This multiple attempt ridiculousness is to accomodate VP7:
// 1. The wrapper can unilaterally reduce the framerate for low bitrates.
// 2. The codec inexplicably likes to reject some frames. Perhaps there
// is a good reason for this...
int encodingAttempts = 0;
double starttime = 0;
double endtime = 0;
while (_lengthEncFrame == 0)
{
starttime = clock()/(double)CLOCKS_PER_SEC;
_inputVideoBuffer.SetWidth(_inst.width);
_inputVideoBuffer.SetHeight(_inst.height);
//_lengthEncFrame = _encoder->Encode(_inputVideoBuffer, _encodedVideoBuffer, _frameInfo,
// _inst.frameRate, _requestKeyFrame && !(_framecnt%50));
endtime = clock()/(double)CLOCKS_PER_SEC;
_encodedVideoBuffer.SetHeight(_inst.height);
_encodedVideoBuffer.SetWidth(_inst.width);
if (_lengthEncFrame < 0)
{
(*_log) << "Error in encoder: " << _lengthEncFrame << std::endl;
fprintf(stderr,"\n\nError in encoder: %d\n\n", _lengthEncFrame);
exit(EXIT_FAILURE);
}
_sumEncBytes += _lengthEncFrame;
encodingAttempts++;
if (encodingAttempts > 50)
{
(*_log) << "Unable to encode frame: " << _framecnt << std::endl;
fprintf(stderr,"\n\nUnable to encode frame: %d\n\n", _framecnt);
exit(EXIT_FAILURE);
}
}
_totalEncodeTime += endtime - starttime;
if (encodingAttempts > 1)
{
(*_log) << encodingAttempts << " attempts required to encode frame: " <<
_framecnt + 1 << std::endl;
fprintf(stderr,"\n%d attempts required to encode frame: %d\n", encodingAttempts,
_framecnt + 1);
}
return false;
}
int
NormalTest::Decode(int lossValue)
{
_encodedVideoBuffer.SetWidth(_inst.width);
_encodedVideoBuffer.SetHeight(_inst.height);
int lengthDecFrame = 0;
//int lengthDecFrame = _decoder->Decode(_encodedVideoBuffer, _decodedVideoBuffer);
//_totalDecodeTime += (double)((clock()/(double)CLOCKS_PER_SEC) - starttime);
if (lengthDecFrame < 0)
{
return lengthDecFrame;
}
_encodedVideoBuffer.SetLength(0);
return lengthDecFrame;
}

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_NORMAL_TEST_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_NORMAL_TEST_H_
#include "test.h"
class NormalTest : public CodecTest
{
public:
NormalTest();
NormalTest(std::string name, std::string description, unsigned int testNo);
NormalTest(std::string name, std::string description, WebRtc_UWord32 bitRate, unsigned int testNo);
virtual ~NormalTest() {};
virtual void Perform();
protected:
virtual void Setup();
virtual void Teardown();
virtual bool Encode();
virtual int Decode(int lossValue = 0);
virtual void CodecSpecific_InitBitrate()=0;
virtual int DoPacketLoss() {return 0;};
FILE* _sourceFile;
FILE* _decodedFile;
FILE* _encodedFile;
double _totalEncodeTime;
double _totalDecodeTime;
unsigned int _framecnt;
bool _requestKeyFrame;
unsigned int _testNo;
int _lengthEncFrame;
bool _appendNext;
};
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_NORMAL_TEST_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "packet_loss_test.h"
#include "video_source.h"
#include <sstream>
#include <cassert>
#include <string.h>
using namespace webrtc;
PacketLossTest::PacketLossTest()
:
NormalAsyncTest("PacketLossTest", "Encode, remove lost packets, decode", 300,
5),
_lossRate(0.1),
_lossProbability(0.1),
_lastFrame(NULL),
_lastFrameLength(0)
{
}
PacketLossTest::PacketLossTest(std::string name, std::string description)
:
NormalAsyncTest(name, description, 300, 5),
_lossRate(0.1),
_lossProbability(0.1),
_lastFrame(NULL),
_lastFrameLength(0)
{
}
PacketLossTest::PacketLossTest(std::string name, std::string description, double lossRate, bool useNack, unsigned int rttFrames /* = 0*/)
:
NormalAsyncTest(name, description, 300, 5, rttFrames),
_lossRate(lossRate),
_lastFrame(NULL),
_lastFrameLength(0)
{
assert(lossRate >= 0 && lossRate <= 1);
if (useNack)
{
_lossProbability = 0;
}
else
{
_lossProbability = lossRate;
}
}
void
PacketLossTest::Encoded(const EncodedImage& encodedImage)
{
// push timestamp to queue
_frameQueue.push_back(encodedImage._timeStamp);
NormalAsyncTest::Encoded(encodedImage);
}
void
PacketLossTest::Decoded(const VideoFrame& decodedImage)
{
// check the frame queue if any frames have gone missing
assert(!_frameQueue.empty()); // decoded frame is not in the queue
while(_frameQueue.front() < decodedImage.TimeStamp())
{
// this frame is missing
// write previous decoded frame again (frame freeze)
if (_decodedFile && _lastFrame)
{
if (fwrite(_lastFrame, 1, _lastFrameLength,
_decodedFile) != _lastFrameLength) {
return;
}
}
// remove frame from queue
_frameQueue.pop_front();
}
// Decoded frame is not in the queue.
assert(_frameQueue.front() == decodedImage.TimeStamp());
// pop the current frame
_frameQueue.pop_front();
// save image for future freeze-frame
if (_lastFrameLength < decodedImage.Length())
{
if (_lastFrame) delete [] _lastFrame;
_lastFrame = new WebRtc_UWord8[decodedImage.Length()];
}
memcpy(_lastFrame, decodedImage.Buffer(), decodedImage.Length());
_lastFrameLength = decodedImage.Length();
NormalAsyncTest::Decoded(decodedImage);
}
void
PacketLossTest::Teardown()
{
if (_totalKept + _totalThrown > 0)
{
printf("Target packet loss rate: %.4f\n", _lossProbability);
printf("Actual packet loss rate: %.4f\n", (_totalThrown * 1.0f) / (_totalKept + _totalThrown));
printf("Channel rate: %.2f kbps\n",
0.001 * 8.0 * _sumChannelBytes / ((_framecnt * 1.0f) / _inst.maxFramerate));
}
else
{
printf("No packet losses inflicted\n");
}
NormalAsyncTest::Teardown();
}
void
PacketLossTest::Setup()
{
const VideoSource source(_inname, _inst.width, _inst.height, _inst.maxFramerate);
std::stringstream ss;
std::string lossRateStr;
ss << _lossRate;
ss >> lossRateStr;
_encodedName = source.GetName() + "-" + lossRateStr;
_outname = "out-" + source.GetName() + "-" + lossRateStr;
if (_lossProbability != _lossRate)
{
_encodedName += "-nack";
_outname += "-nack";
}
_encodedName += ".vp8";
_outname += ".yuv";
_totalKept = 0;
_totalThrown = 0;
_sumChannelBytes = 0;
NormalAsyncTest::Setup();
}
void
PacketLossTest::CodecSpecific_InitBitrate()
{
assert(_bitRate > 0);
WebRtc_UWord32 simulatedBitRate;
if (_lossProbability != _lossRate)
{
// Simulating NACK
simulatedBitRate = WebRtc_UWord32(_bitRate / (1 + _lossRate));
}
else
{
simulatedBitRate = _bitRate;
}
int rtt = 0;
if (_inst.maxFramerate > 0)
rtt = _rttFrames * (1000 / _inst.maxFramerate);
_encoder->SetChannelParameters((WebRtc_UWord32)(_lossProbability * 255.0),
rtt);
_encoder->SetRates(simulatedBitRate, _inst.maxFramerate);
}
int PacketLossTest::DoPacketLoss()
{
// Only packet loss for delta frames
// TODO(mikhal): Identify delta frames
// First frame so never a delta frame.
if (_frameToDecode->_frame->Length() == 0 || _sumChannelBytes == 0)
{
_sumChannelBytes += _frameToDecode->_frame->Length();
return 0;
}
unsigned char *packet = NULL;
VideoFrame newEncBuf;
newEncBuf.VerifyAndAllocate(_lengthSourceFrame);
_inBufIdx = 0;
_outBufIdx = 0;
int size = 1;
int kept = 0;
int thrown = 0;
while ((size = NextPacket(1500, &packet)) > 0)
{
if (!PacketLoss(_lossProbability, thrown))
{
InsertPacket(&newEncBuf, packet, size);
kept++;
}
else
{
// Use the ByteLoss function if you want to lose only
// parts of a packet, and not the whole packet.
//int size2 = ByteLoss(size, packet, 15);
thrown++;
//if (size2 != size)
//{
// InsertPacket(&newEncBuf, packet, size2);
//}
}
}
int lossResult = (thrown!=0); // 0 = no loss 1 = loss(es)
if (lossResult)
{
lossResult += (kept==0); // 2 = all lost = full frame
}
_frameToDecode->_frame->CopyFrame(newEncBuf.Length(), newEncBuf.Buffer());
_sumChannelBytes += newEncBuf.Length();
_totalKept += kept;
_totalThrown += thrown;
return lossResult;
//printf("Threw away: %d out of %d packets\n", thrown, thrown + kept);
//printf("Encoded left: %d bytes\n", _encodedVideoBuffer.Length());
}
int PacketLossTest::NextPacket(int mtu, unsigned char **pkg)
{
unsigned char *buf = _frameToDecode->_frame->Buffer();
*pkg = buf + _inBufIdx;
if (static_cast<long>(_frameToDecode->_frame->Length()) - _inBufIdx <= mtu)
{
int size = _frameToDecode->_frame->Length() - _inBufIdx;
_inBufIdx = _frameToDecode->_frame->Length();
return size;
}
_inBufIdx += mtu;
return mtu;
}
int PacketLossTest::ByteLoss(int size, unsigned char *pkg, int bytesToLose)
{
return size;
}
void PacketLossTest::InsertPacket(VideoFrame *buf, unsigned char *pkg, int size)
{
if (static_cast<long>(buf->Size()) - _outBufIdx < size)
{
printf("InsertPacket error!\n");
return;
}
memcpy(buf->Buffer() + _outBufIdx, pkg, size);
buf->SetLength(buf->Length() + size);
_outBufIdx += size;
}

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_PACKET_LOSS_TEST_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_PACKET_LOSS_TEST_H_
#include <list>
#include "normal_async_test.h"
class PacketLossTest : public NormalAsyncTest
{
public:
PacketLossTest();
virtual ~PacketLossTest() {if(_lastFrame) {delete [] _lastFrame; _lastFrame = NULL;}}
virtual void Encoded(const webrtc::EncodedImage& encodedImage);
virtual void Decoded(const webrtc::VideoFrame& decodedImage);
protected:
PacketLossTest(std::string name, std::string description);
PacketLossTest(std::string name,
std::string description,
double lossRate,
bool useNack,
unsigned int rttFrames = 0);
virtual void Setup();
virtual void Teardown();
virtual void CodecSpecific_InitBitrate();
virtual int DoPacketLoss();
virtual int NextPacket(int size, unsigned char **pkg);
virtual int ByteLoss(int size, unsigned char *pkg, int bytesToLose);
virtual void InsertPacket(webrtc::VideoFrame *buf, unsigned char *pkg,
int size);
int _inBufIdx;
int _outBufIdx;
// When NACK is being simulated _lossProbabilty is zero,
// otherwise it is set equal to _lossRate.
// Desired channel loss rate.
double _lossRate;
// Probability used to simulate packet drops.
double _lossProbability;
int _totalKept;
int _totalThrown;
int _sumChannelBytes;
std::list<WebRtc_UWord32> _frameQueue;
WebRtc_UWord8* _lastFrame;
WebRtc_UWord32 _lastFrameLength;
};
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_PACKET_LOSS_TEST_H_

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function plotBenchmark(fileNames, export)
%PLOTBENCHMARK Plots and exports video codec benchmarking results.
% PLOTBENCHMARK(FILENAMES, EXPORT) parses the video codec benchmarking result
% files given by the cell array of strings FILENAME. It plots the results and
% optionally exports each plot to an appropriately named file.
%
% EXPORT parameter:
% 'none' No file exports.
% 'eps' Exports to eps files (default).
% 'pdf' Exports to eps files and uses the command-line utility
% epstopdf to obtain pdf files.
%
% Example:
% plotBenchmark({'H264Benchmark.txt' 'LSVXBenchmark.txt'}, 'pdf')
if (nargin < 1)
error('Too few input arguments');
elseif (nargin < 2)
export = 'eps';
end
if ~iscell(fileNames)
if ischar(fileNames)
% one single file name as a string is ok
if size(fileNames,1) > 1
% this is a char matrix, not ok
error('First argument must not be a char matrix');
end
% wrap in a cell array
fileNames = {fileNames};
else
error('First argument must be a cell array of strings');
end
end
if ~ischar(export)
error('Second argument must be a string');
end
outpath = 'BenchmarkPlots';
[status, errMsg] = mkdir(outpath);
if status == 0
error(errMsg);
end
nCases = 0;
testCases = [];
% Read each test result file
for fileIdx = 1:length(fileNames)
if ~isstr(fileNames{fileIdx})
error('First argument must be a cell array of strings');
end
fid = fopen(fileNames{fileIdx}, 'rt');
if fid == -1
error(['Unable to open ' fileNames{fileIdx}]);
end
version = '1.0';
if ~strcmp(fgetl(fid), ['#!benchmark' version])
fclose(fid);
error(['Requires benchmark file format version ' version]);
end
% Parse results file into testCases struct
codec = fgetl(fid);
tline = fgetl(fid);
while(tline ~= -1)
nCases = nCases + 1;
delim = strfind(tline, ',');
name = tline(1:delim(1)-1);
% Drop underscored suffix from name
underscore = strfind(name, '_');
if ~isempty(underscore)
name = name(1:underscore(1)-1);
end
resolution = tline(delim(1)+1:delim(2)-1);
frameRate = tline(delim(2)+1:end);
tline = fgetl(fid);
delim = strfind(tline, ',');
bitrateLabel = tline(1:delim(1)-1);
bitrate = sscanf(tline(delim(1):end),',%f');
tline = fgetl(fid);
delim = strfind(tline, ',');
psnrLabel = tline(1:delim(1)-1);
psnr = sscanf(tline(delim(1):end),',%f');
% Default data for the optional lines
speedLabel = 'Default';
speed = 0;
ssimLabel = 'Default';
ssim = 0;
tline = fgetl(fid);
delim = strfind(tline, ',');
while ~isempty(delim)
% More data
% Check type of data
if strncmp(lower(tline), 'speed', 5)
% Speed data included
speedLabel = tline(1:delim(1)-1);
speed = sscanf(tline(delim(1):end), ',%f');
tline = fgetl(fid);
elseif strncmp(lower(tline), 'encode time', 11)
% Encode and decode times included
% TODO: take care of the data
% pop two lines from file
tline = fgetl(fid);
tline = fgetl(fid);
elseif strncmp(tline, 'SSIM', 4)
% SSIM data included
ssimLabel = tline(1:delim(1)-1);
ssim = sscanf(tline(delim(1):end), ',%f');
tline = fgetl(fid);
end
delim = strfind(tline, ',');
end
testCases = [testCases struct('codec', codec, 'name', name, 'resolution', ...
resolution, 'frameRate', frameRate, 'bitrate', bitrate, 'psnr', psnr, ...
'speed', speed, 'bitrateLabel', bitrateLabel, 'psnrLabel', psnrLabel, ...
'speedLabel', speedLabel, ...
'ssim', ssim, 'ssimLabel', ssimLabel)];
tline = fgetl(fid);
end
fclose(fid);
end
i = 0;
casesPsnr = testCases;
while ~isempty(casesPsnr)
i = i + 1;
casesPsnr = plotOnePsnr(casesPsnr, i, export, outpath);
end
casesSSIM = testCases;
while ~isempty(casesSSIM)
i = i + 1;
casesSSIM = plotOneSSIM(casesSSIM, i, export, outpath);
end
casesSpeed = testCases;
while ~isempty(casesSpeed)
if casesSpeed(1).speed == 0
casesSpeed = casesSpeed(2:end);
else
i = i + 1;
casesSpeed = plotOneSpeed(casesSpeed, i, export, outpath);
end
end
%%%%%%%%%%%%%%%%%%
%% SUBFUNCTIONS %%
%%%%%%%%%%%%%%%%%%
function casesOut = plotOnePsnr(cases, num, export, outpath)
% Find matching specs
plotIdx = 1;
for i = 2:length(cases)
if strcmp(cases(1).resolution, cases(i).resolution) & ...
strcmp(cases(1).frameRate, cases(i).frameRate)
plotIdx = [plotIdx i];
end
end
% Return unplotted cases
casesOut = cases(setdiff(1:length(cases), plotIdx));
cases = cases(plotIdx);
% Prune similar results
for i = 1:length(cases)
simIndx = find(abs(cases(i).bitrate - [cases(i).bitrate(2:end) ; 0]) < 10);
while ~isempty(simIndx)
diffIndx = setdiff(1:length(cases(i).bitrate), simIndx);
cases(i).psnr = cases(i).psnr(diffIndx);
cases(i).bitrate = cases(i).bitrate(diffIndx);
simIndx = find(abs(cases(i).bitrate - [cases(i).bitrate(2:end) ; 0]) < 10);
end
end
% Prepare figure with axis labels and so on
hFig = figure(num);
clf;
hold on;
grid on;
axis([0 1100 20 50]);
set(gca, 'XTick', 0:200:1000);
set(gca, 'YTick', 20:10:60);
xlabel(cases(1).bitrateLabel);
ylabel(cases(1).psnrLabel);
res = cases(1).resolution;
frRate = cases(1).frameRate;
title([res ', ' frRate]);
hLines = [];
codecs = {};
sequences = {};
i = 0;
while ~isempty(cases)
i = i + 1;
[cases, hLine, codec, sequences] = plotOneCodec(cases, 'bitrate', 'psnr', i, sequences, 1);
% Stored to generate the legend
hLines = [hLines ; hLine];
codecs = {codecs{:} codec};
end
legend(hLines, codecs, 4);
hold off;
if ~strcmp(export, 'none')
% Export figure to an eps file
res = stripws(res);
frRate = stripws(frRate);
exportName = [outpath '/psnr-' res '-' frRate];
exportfig(hFig, exportName, 'Format', 'eps2', 'Color', 'cmyk');
end
if strcmp(export, 'pdf')
% Use the epstopdf utility to convert to pdf
system(['epstopdf ' exportName '.eps']);
end
function casesOut = plotOneSSIM(cases, num, export, outpath)
% Find matching specs
plotIdx = 1;
for i = 2:length(cases)
if strcmp(cases(1).resolution, cases(i).resolution) & ...
strcmp(cases(1).frameRate, cases(i).frameRate)
plotIdx = [plotIdx i];
end
end
% Return unplotted cases
casesOut = cases(setdiff(1:length(cases), plotIdx));
cases = cases(plotIdx);
% Prune similar results
for i = 1:length(cases)
simIndx = find(abs(cases(i).bitrate - [cases(i).bitrate(2:end) ; 0]) < 10);
while ~isempty(simIndx)
diffIndx = setdiff(1:length(cases(i).bitrate), simIndx);
cases(i).ssim = cases(i).ssim(diffIndx);
cases(i).bitrate = cases(i).bitrate(diffIndx);
simIndx = find(abs(cases(i).bitrate - [cases(i).bitrate(2:end) ; 0]) < 10);
end
end
% Prepare figure with axis labels and so on
hFig = figure(num);
clf;
hold on;
grid on;
axis([0 1100 0.5 1]); % y-limit are set to 'auto' below
set(gca, 'XTick', 0:200:1000);
%set(gca, 'YTick', 20:10:60);
xlabel(cases(1).bitrateLabel);
ylabel(cases(1).ssimLabel);
res = cases(1).resolution;
frRate = cases(1).frameRate;
title([res ', ' frRate]);
hLines = [];
codecs = {};
sequences = {};
i = 0;
while ~isempty(cases)
i = i + 1;
[cases, hLine, codec, sequences] = plotOneCodec(cases, 'bitrate', 'ssim', i, sequences, 1);
% Stored to generate the legend
hLines = [hLines ; hLine];
codecs = {codecs{:} codec};
end
%set(gca,'YLimMode','auto')
set(gca,'YLim',[0.5 1])
set(gca,'YScale','log')
legend(hLines, codecs, 4);
hold off;
if ~strcmp(export, 'none')
% Export figure to an eps file
res = stripws(res);
frRate = stripws(frRate);
exportName = [outpath '/psnr-' res '-' frRate];
exportfig(hFig, exportName, 'Format', 'eps2', 'Color', 'cmyk');
end
if strcmp(export, 'pdf')
% Use the epstopdf utility to convert to pdf
system(['epstopdf ' exportName '.eps']);
end
function casesOut = plotOneSpeed(cases, num, export, outpath)
% Find matching specs
plotIdx = 1;
for i = 2:length(cases)
if strcmp(cases(1).resolution, cases(i).resolution) & ...
strcmp(cases(1).frameRate, cases(i).frameRate) & ...
strcmp(cases(1).name, cases(i).name)
plotIdx = [plotIdx i];
end
end
% Return unplotted cases
casesOut = cases(setdiff(1:length(cases), plotIdx));
cases = cases(plotIdx);
% Prune similar results
for i = 1:length(cases)
simIndx = find(abs(cases(i).psnr - [cases(i).psnr(2:end) ; 0]) < 0.25);
while ~isempty(simIndx)
diffIndx = setdiff(1:length(cases(i).psnr), simIndx);
cases(i).psnr = cases(i).psnr(diffIndx);
cases(i).speed = cases(i).speed(diffIndx);
simIndx = find(abs(cases(i).psnr - [cases(i).psnr(2:end) ; 0]) < 0.25);
end
end
hFig = figure(num);
clf;
hold on;
%grid on;
xlabel(cases(1).psnrLabel);
ylabel(cases(1).speedLabel);
res = cases(1).resolution;
name = cases(1).name;
frRate = cases(1).frameRate;
title([name ', ' res ', ' frRate]);
hLines = [];
codecs = {};
sequences = {};
i = 0;
while ~isempty(cases)
i = i + 1;
[cases, hLine, codec, sequences] = plotOneCodec(cases, 'psnr', 'speed', i, sequences, 0);
% Stored to generate the legend
hLines = [hLines ; hLine];
codecs = {codecs{:} codec};
end
legend(hLines, codecs, 1);
hold off;
if ~strcmp(export, 'none')
% Export figure to an eps file
res = stripws(res);
frRate = stripws(frRate);
exportName = [outpath '/speed-' name '-' res '-' frRate];
exportfig(hFig, exportName, 'Format', 'eps2', 'Color', 'cmyk');
end
if strcmp(export, 'pdf')
% Use the epstopdf utility to convert to pdf
system(['epstopdf ' exportName '.eps']);
end
function [casesOut, hLine, codec, sequences] = plotOneCodec(cases, xfield, yfield, num, sequences, annotatePlot)
plotStr = {'gx-', 'bo-', 'r^-', 'kd-', 'cx-', 'go--', 'b^--'};
% Find matching codecs
plotIdx = 1;
for i = 2:length(cases)
if strcmp(cases(1).codec, cases(i).codec)
plotIdx = [plotIdx i];
end
end
% Return unplotted cases
casesOut = cases(setdiff(1:length(cases), plotIdx));
cases = cases(plotIdx);
for i = 1:length(cases)
% Plot a single case
hLine = plot(getfield(cases(i), xfield), getfield(cases(i), yfield), plotStr{num}, ...
'LineWidth', 1.1, 'MarkerSize', 6);
end
% hLine handle and codec are returned to construct the legend afterwards
codec = cases(1).codec;
if annotatePlot == 0
return;
end
for i = 1:length(cases)
% Print the codec name as a text label
% Ensure each codec is only printed once
sequencePlotted = 0;
for j = 1:length(sequences)
if strcmp(cases(i).name, sequences{j})
sequencePlotted = 1;
break;
end
end
if sequencePlotted == 0
text(getfield(cases(i), xfield, {1}), getfield(cases(i), yfield, {1}), ...
[' ' cases(i).name]);
sequences = {sequences{:} cases(i).name};
end
end
% Strip whitespace from string
function str = stripws(str)
if ~isstr(str)
error('String required');
end
str = str(setdiff(1:length(str), find(isspace(str) == 1)));

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "test.h"
#include <cstring>
#include <iostream>
#include "testsupport/metrics/video_metrics.h"
using namespace webrtc;
long filesize(const char *filename); // local function defined at end of file
CodecTest::CodecTest(std::string name, std::string description)
:
_bitRate(0),
_inname(""),
_outname(""),
_encodedName(""),
_name(name),
_description(description)
{
memset(&_inst, 0, sizeof(_inst));
unsigned int seed = static_cast<unsigned int>(0);
std::srand(seed);
}
CodecTest::CodecTest(std::string name, std::string description,
WebRtc_UWord32 bitRate)
:
_bitRate(bitRate),
_inname(""),
_outname(""),
_encodedName(""),
_name(name),
_description(description)
{
memset(&_inst, 0, sizeof(_inst));
unsigned int seed = static_cast<unsigned int>(0);
std::srand(seed);
}
void
CodecTest::Print()
{
std::cout << _name << " completed!" << std::endl;
(*_log) << _name << std::endl;
(*_log) << _description << std::endl;
(*_log) << "Input file: " << _inname << std::endl;
(*_log) << "Output file: " << _outname << std::endl;
webrtc::test::QualityMetricsResult psnr;
webrtc::test::QualityMetricsResult ssim;
I420PSNRFromFiles(_inname.c_str(), _outname.c_str(), _inst.width,
_inst.height, &psnr);
I420SSIMFromFiles(_inname.c_str(), _outname.c_str(), _inst.width,
_inst.height, &ssim);
(*_log) << "PSNR: " << psnr.average << std::endl;
std::cout << "PSNR: " << psnr.average << std::endl << std::endl;
(*_log) << "SSIM: " << ssim.average << std::endl;
std::cout << "SSIM: " << ssim.average << std::endl << std::endl;
(*_log) << std::endl;
}
void
CodecTest::Setup()
{
int widhei = _inst.width*_inst.height;
_lengthSourceFrame = 3*widhei/2;
_sourceBuffer = new unsigned char[_lengthSourceFrame];
}
void
CodecTest::CodecSettings(int width, int height,
WebRtc_UWord32 frameRate /*=30*/,
WebRtc_UWord32 bitRate /*=0*/)
{
if (bitRate > 0)
{
_bitRate = bitRate;
}
else if (_bitRate == 0)
{
_bitRate = 600;
}
_inst.codecType = kVideoCodecVP8;
_inst.codecSpecific.VP8.feedbackModeOn = true;
_inst.maxFramerate = (unsigned char)frameRate;
_inst.startBitrate = (int)_bitRate;
_inst.maxBitrate = 8000;
_inst.width = width;
_inst.height = height;
}
void
CodecTest::Teardown()
{
delete [] _sourceBuffer;
}
void
CodecTest::SetEncoder(webrtc::VideoEncoder*encoder)
{
_encoder = encoder;
}
void
CodecTest::SetDecoder(VideoDecoder*decoder)
{
_decoder = decoder;
}
void
CodecTest::SetLog(std::fstream* log)
{
_log = log;
}
double CodecTest::ActualBitRate(int nFrames)
{
return 8.0 * _sumEncBytes / (nFrames / _inst.maxFramerate);
}
bool CodecTest::PacketLoss(double lossRate, int /*thrown*/)
{
return RandUniform() < lossRate;
}
void
CodecTest::VideoEncodedBufferToEncodedImage(VideoFrame& videoBuffer,
EncodedImage &image)
{
image._buffer = videoBuffer.Buffer();
image._length = videoBuffer.Length();
image._size = videoBuffer.Size();
//image._frameType = static_cast<VideoFrameType>
// (videoBuffer.GetFrameType());
image._timeStamp = videoBuffer.TimeStamp();
image._encodedWidth = videoBuffer.Width();
image._encodedHeight = videoBuffer.Height();
image._completeFrame = true;
}
long filesize(const char *filename)
{
FILE *f = fopen(filename,"rb"); /* open the file in read only */
long size = 0;
if (fseek(f,0,SEEK_END)==0) /* seek was successful */
size = ftell(f);
fclose(f);
return size;
}

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAWEWORK_TEST_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAWEWORK_TEST_H_
#include "modules/interface/module_common_types.h"
#include "video_codec_interface.h"
#include <string>
#include <fstream>
#include <cstdlib>
class CodecTest
{
public:
CodecTest(std::string name, std::string description);
CodecTest(std::string name, std::string description,
WebRtc_UWord32 bitRate);
virtual ~CodecTest() {};
virtual void Perform()=0;
virtual void Print();
void SetEncoder(webrtc::VideoEncoder *encoder);
void SetDecoder(webrtc::VideoDecoder *decoder);
void SetLog(std::fstream* log);
protected:
virtual void Setup();
virtual void CodecSettings(int width,
int height,
WebRtc_UWord32 frameRate=30,
WebRtc_UWord32 bitRate=0);
virtual void Teardown();
double ActualBitRate(int nFrames);
virtual bool PacketLoss(double lossRate, int /*thrown*/);
static double RandUniform() { return (std::rand() + 1.0)/(RAND_MAX + 1.0); }
static void VideoEncodedBufferToEncodedImage(
webrtc::VideoFrame& videoBuffer,
webrtc::EncodedImage &image);
webrtc::VideoEncoder* _encoder;
webrtc::VideoDecoder* _decoder;
WebRtc_UWord32 _bitRate;
unsigned int _lengthSourceFrame;
unsigned char* _sourceBuffer;
webrtc::VideoFrame _inputVideoBuffer;
// TODO(mikhal): For now using VideoFrame for encodedBuffer, should use a
// designated class.
webrtc::VideoFrame _encodedVideoBuffer;
webrtc::VideoFrame _decodedVideoBuffer;
webrtc::VideoCodec _inst;
std::fstream* _log;
std::string _inname;
std::string _outname;
std::string _encodedName;
int _sumEncBytes;
private:
std::string _name;
std::string _description;
};
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAWEWORK_TEST_H_

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# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'conditions': [
['include_tests==1', {
'targets': [
{
'target_name': 'test_framework',
'type': '<(library)',
'dependencies': [
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/common_video/common_video.gyp:common_video',
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/test/metrics.gyp:metrics',
'<(webrtc_root)/test/test.gyp:test_support',
],
'include_dirs': [
'../interface',
'<(DEPTH)/testing/gtest/include',
'../../../../common_video/interface',
],
'direct_dependent_settings': {
'include_dirs': [
'../interface',
],
},
'sources': [
# header files
'benchmark.h',
'normal_async_test.h',
'normal_test.h',
'packet_loss_test.h',
'test.h',
'unit_test.h',
'video_source.h',
# source files
'benchmark.cc',
'normal_async_test.cc',
'normal_test.cc',
'packet_loss_test.cc',
'test.cc',
'unit_test.cc',
'video_source.cc',
],
},
], # targets
}], # include_tests
], # conditions
}

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <math.h>
#include <stdlib.h>
#include <string.h>
#include <cassert>
#include "gtest/gtest.h"
#include "testsupport/fileutils.h"
#include "tick_util.h"
#include "unit_test.h"
#include "video_source.h"
using namespace webrtc;
UnitTest::UnitTest()
:
CodecTest("UnitTest", "Unit test"),
_tests(0),
_errors(0),
_source(NULL),
_refFrame(NULL),
_refEncFrame(NULL),
_refDecFrame(NULL),
_refEncFrameLength(0),
_sourceFile(NULL),
_encodeCompleteCallback(NULL),
_decodeCompleteCallback(NULL)
{
}
UnitTest::UnitTest(std::string name, std::string description)
:
CodecTest(name, description),
_tests(0),
_errors(0),
_source(NULL),
_refFrame(NULL),
_refEncFrame(NULL),
_refDecFrame(NULL),
_refEncFrameLength(0),
_sourceFile(NULL),
_encodeCompleteCallback(NULL),
_decodeCompleteCallback(NULL)
{
}
UnitTest::~UnitTest()
{
if (_encodeCompleteCallback) {
delete _encodeCompleteCallback;
}
if (_decodeCompleteCallback) {
delete _decodeCompleteCallback;
}
if (_source) {
delete _source;
}
if (_refFrame) {
delete [] _refFrame;
}
if (_refDecFrame) {
delete [] _refDecFrame;
}
if (_sourceBuffer) {
delete [] _sourceBuffer;
}
if (_sourceFile) {
fclose(_sourceFile);
}
if (_refEncFrame) {
delete [] _refEncFrame;
}
}
WebRtc_Word32
UnitTestEncodeCompleteCallback::Encoded(EncodedImage& encodedImage,
const webrtc::CodecSpecificInfo* codecSpecificInfo,
const webrtc::RTPFragmentationHeader*
fragmentation)
{
_encodedVideoBuffer->VerifyAndAllocate(encodedImage._size);
_encodedVideoBuffer->CopyFrame(encodedImage._size, encodedImage._buffer);
_encodedVideoBuffer->SetLength(encodedImage._length);
// _encodedVideoBuffer->SetFrameType(encodedImage._frameType);
_encodedVideoBuffer->SetWidth(
(WebRtc_UWord16)encodedImage._encodedWidth);
_encodedVideoBuffer->SetHeight(
(WebRtc_UWord16)encodedImage._encodedHeight);
_encodedVideoBuffer->SetTimeStamp(encodedImage._timeStamp);
_encodeComplete = true;
_encodedFrameType = encodedImage._frameType;
return 0;
}
WebRtc_Word32 UnitTestDecodeCompleteCallback::Decoded(VideoFrame& image)
{
_decodedVideoBuffer->CopyFrame(image.Length(), image.Buffer());
_decodedVideoBuffer->SetWidth(image.Width());
_decodedVideoBuffer->SetHeight(image.Height());
_decodedVideoBuffer->SetTimeStamp(image.TimeStamp());
_decodeComplete = true;
return 0;
}
bool
UnitTestEncodeCompleteCallback::EncodeComplete()
{
if (_encodeComplete)
{
_encodeComplete = false;
return true;
}
return false;
}
VideoFrameType
UnitTestEncodeCompleteCallback::EncodedFrameType() const
{
return _encodedFrameType;
}
bool
UnitTestDecodeCompleteCallback::DecodeComplete()
{
if (_decodeComplete)
{
_decodeComplete = false;
return true;
}
return false;
}
WebRtc_UWord32
UnitTest::WaitForEncodedFrame() const
{
WebRtc_Word64 startTime = TickTime::MillisecondTimestamp();
while (TickTime::MillisecondTimestamp() - startTime < kMaxWaitEncTimeMs)
{
if (_encodeCompleteCallback->EncodeComplete())
{
return _encodedVideoBuffer.Length();
}
}
return 0;
}
WebRtc_UWord32
UnitTest::WaitForDecodedFrame() const
{
WebRtc_Word64 startTime = TickTime::MillisecondTimestamp();
while (TickTime::MillisecondTimestamp() - startTime < kMaxWaitDecTimeMs)
{
if (_decodeCompleteCallback->DecodeComplete())
{
return _decodedVideoBuffer.Length();
}
}
return 0;
}
WebRtc_UWord32
UnitTest::CodecSpecific_SetBitrate(WebRtc_UWord32 bitRate,
WebRtc_UWord32 /* frameRate */)
{
return _encoder->SetRates(bitRate, _inst.maxFramerate);
}
void
UnitTest::Setup()
{
// Use _sourceFile as a check to prevent multiple Setup() calls.
if (_sourceFile != NULL)
{
return;
}
if (_encodeCompleteCallback == NULL)
{
_encodeCompleteCallback =
new UnitTestEncodeCompleteCallback(&_encodedVideoBuffer);
}
if (_decodeCompleteCallback == NULL)
{
_decodeCompleteCallback =
new UnitTestDecodeCompleteCallback(&_decodedVideoBuffer);
}
_encoder->RegisterEncodeCompleteCallback(_encodeCompleteCallback);
_decoder->RegisterDecodeCompleteCallback(_decodeCompleteCallback);
_source = new VideoSource(webrtc::test::ProjectRootPath() +
"resources/foreman_cif.yuv", kCIF);
_lengthSourceFrame = _source->GetFrameLength();
_refFrame = new unsigned char[_lengthSourceFrame];
_refDecFrame = new unsigned char[_lengthSourceFrame];
_sourceBuffer = new unsigned char [_lengthSourceFrame];
_sourceFile = fopen(_source->GetFileName().c_str(), "rb");
ASSERT_TRUE(_sourceFile != NULL);
_inst.maxFramerate = _source->GetFrameRate();
_bitRate = 300;
_inst.startBitrate = 300;
_inst.maxBitrate = 4000;
_inst.width = _source->GetWidth();
_inst.height = _source->GetHeight();
_inst.codecSpecific.VP8.denoisingOn = true;
// Get input frame.
_inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
ASSERT_TRUE(fread(_refFrame, 1, _lengthSourceFrame, _sourceFile)
== _lengthSourceFrame);
_inputVideoBuffer.CopyFrame(_lengthSourceFrame, _refFrame);
_inputVideoBuffer.SetWidth(_source->GetWidth());
_inputVideoBuffer.SetHeight(_source->GetHeight());
rewind(_sourceFile);
// Get a reference encoded frame.
_encodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
// Ensures our initial parameters are valid.
EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
_encoder->Encode(_inputVideoBuffer, NULL, NULL);
_refEncFrameLength = WaitForEncodedFrame();
ASSERT_TRUE(_refEncFrameLength > 0);
_refEncFrame = new unsigned char[_refEncFrameLength];
memcpy(_refEncFrame, _encodedVideoBuffer.Buffer(), _refEncFrameLength);
// Get a reference decoded frame.
_decodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
EXPECT_TRUE(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
ASSERT_FALSE(SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK);
unsigned int frameLength = 0;
int i=0;
while (frameLength == 0)
{
if (i > 0)
{
// Insert yet another frame
_inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
ASSERT_TRUE(fread(_refFrame, 1, _lengthSourceFrame,
_sourceFile) == _lengthSourceFrame);
_inputVideoBuffer.CopyFrame(_lengthSourceFrame, _refFrame);
_inputVideoBuffer.SetWidth(_source->GetWidth());
_inputVideoBuffer.SetHeight(_source->GetHeight());
_encoder->Encode(_inputVideoBuffer, NULL, NULL);
ASSERT_TRUE(WaitForEncodedFrame() > 0);
}
EncodedImage encodedImage;
VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
ASSERT_TRUE(_decoder->Decode(encodedImage, 0, NULL)
== WEBRTC_VIDEO_CODEC_OK);
frameLength = WaitForDecodedFrame();
_encodedVideoBuffer.SetLength(0);
i++;
}
rewind(_sourceFile);
EXPECT_TRUE(frameLength == _lengthSourceFrame);
memcpy(_refDecFrame, _decodedVideoBuffer.Buffer(), _lengthSourceFrame);
}
void
UnitTest::Teardown()
{
// Use _sourceFile as a check to prevent multiple Teardown() calls.
if (_sourceFile == NULL)
{
return;
}
_encoder->Release();
_decoder->Release();
fclose(_sourceFile);
_sourceFile = NULL;
delete [] _refFrame;
_refFrame = NULL;
delete [] _refEncFrame;
_refEncFrame = NULL;
delete [] _refDecFrame;
_refDecFrame = NULL;
delete [] _sourceBuffer;
_sourceBuffer = NULL;
}
void
UnitTest::Print()
{
}
int
UnitTest::DecodeWithoutAssert()
{
EncodedImage encodedImage;
VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
int ret = _decoder->Decode(encodedImage, 0, NULL);
int frameLength = WaitForDecodedFrame();
_encodedVideoBuffer.SetLength(0);
return ret == WEBRTC_VIDEO_CODEC_OK ? frameLength : ret;
}
int
UnitTest::Decode()
{
EncodedImage encodedImage;
VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
if (encodedImage._length == 0)
{
return WEBRTC_VIDEO_CODEC_OK;
}
int ret = _decoder->Decode(encodedImage, 0, NULL);
unsigned int frameLength = WaitForDecodedFrame();
assert(ret == WEBRTC_VIDEO_CODEC_OK && (frameLength == 0 || frameLength
== _lengthSourceFrame));
EXPECT_TRUE(ret == WEBRTC_VIDEO_CODEC_OK && (frameLength == 0 || frameLength
== _lengthSourceFrame));
_encodedVideoBuffer.SetLength(0);
return ret == WEBRTC_VIDEO_CODEC_OK ? frameLength : ret;
}
// Test pure virtual VideoEncoder and VideoDecoder APIs.
void
UnitTest::Perform()
{
UnitTest::Setup();
int frameLength;
VideoFrame inputImage;
EncodedImage encodedImage;
//----- Encoder parameter tests -----
//-- Calls before InitEncode() --
// We want to revert the initialization done in Setup().
EXPECT_TRUE(_encoder->Release() == WEBRTC_VIDEO_CODEC_OK);
EXPECT_TRUE(_encoder->Encode(_inputVideoBuffer, NULL, NULL)
== WEBRTC_VIDEO_CODEC_UNINITIALIZED);
//-- InitEncode() errors --
// Null pointer.
EXPECT_TRUE(_encoder->InitEncode(NULL, 1, 1440) ==
WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
// bit rate exceeds max bit rate
WebRtc_Word32 tmpBitRate = _inst.startBitrate;
WebRtc_Word32 tmpMaxBitRate = _inst.maxBitrate;
_inst.startBitrate = 4000;
_inst.maxBitrate = 3000;
EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) ==
WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
_inst.startBitrate = tmpBitRate;
_inst.maxBitrate = tmpMaxBitRate; //unspecified value
// Bad framerate.
_inst.maxFramerate = 0;
EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) ==
WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
// Seems like we should allow any framerate in range [0, 255].
//_inst.frameRate = 100;
//EXPECT_TRUE(_encoder->InitEncode(&_inst, 1) == -1); // FAILS
_inst.maxFramerate = 30;
// Bad bitrate.
_inst.startBitrate = -1;
EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) ==
WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
_inst.maxBitrate = _inst.startBitrate - 1;
EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) ==
WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
_inst.maxBitrate = 0;
_inst.startBitrate = 300;
// Bad maxBitRate.
_inst.maxBitrate = 200;
EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) ==
WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
_inst.maxBitrate = 4000;
// Bad width.
_inst.width = 0;
EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) < 0);
_inst.width = _source->GetWidth();
// Bad height.
_inst.height = 0;
EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) < 0);
_inst.height = _source->GetHeight();
// Bad number of cores.
EXPECT_TRUE(_encoder->InitEncode(&_inst, -1, 1440) ==
WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
//-- Encode() errors --
// inputVideoBuffer unallocated.
_inputVideoBuffer.Free();
inputImage.Free();
EXPECT_TRUE(_encoder->Encode(inputImage, NULL, NULL) ==
WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
_inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
_inputVideoBuffer.CopyFrame(_lengthSourceFrame, _refFrame);
_inputVideoBuffer.SetWidth(_source->GetWidth());
_inputVideoBuffer.SetHeight(_source->GetHeight());
//----- Encoder stress tests -----
// Vary frame rate and I-frame request.
for (int i = 1; i <= 60; i++)
{
VideoFrameType frame_type = !(i % 2) ? kKeyFrame : kDeltaFrame;
std::vector<VideoFrameType> frame_types(1, frame_type);
EXPECT_TRUE(_encoder->Encode(_inputVideoBuffer, NULL, &frame_types) ==
WEBRTC_VIDEO_CODEC_OK);
EXPECT_TRUE(WaitForEncodedFrame() > 0);
}
// Init then encode.
_encodedVideoBuffer.SetLength(0);
EXPECT_TRUE(_encoder->Encode(_inputVideoBuffer, NULL, NULL) ==
WEBRTC_VIDEO_CODEC_OK);
EXPECT_TRUE(WaitForEncodedFrame() > 0);
EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
_encoder->Encode(_inputVideoBuffer, NULL, NULL);
frameLength = WaitForEncodedFrame();
EXPECT_TRUE(frameLength > 0);
EXPECT_TRUE(CheckIfBitExact(_refEncFrame, _refEncFrameLength,
_encodedVideoBuffer.Buffer(), frameLength) == true);
// Reset then encode.
_encodedVideoBuffer.SetLength(0);
EXPECT_TRUE(_encoder->Encode(_inputVideoBuffer, NULL, NULL) ==
WEBRTC_VIDEO_CODEC_OK);
WaitForEncodedFrame();
EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
_encoder->Encode(_inputVideoBuffer, NULL, NULL);
frameLength = WaitForEncodedFrame();
EXPECT_TRUE(frameLength > 0);
EXPECT_TRUE(CheckIfBitExact(_refEncFrame, _refEncFrameLength,
_encodedVideoBuffer.Buffer(), frameLength) == true);
// Release then encode.
_encodedVideoBuffer.SetLength(0);
EXPECT_TRUE(_encoder->Encode(_inputVideoBuffer, NULL, NULL) ==
WEBRTC_VIDEO_CODEC_OK);
WaitForEncodedFrame();
EXPECT_TRUE(_encoder->Release() == WEBRTC_VIDEO_CODEC_OK);
EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
_encoder->Encode(_inputVideoBuffer, NULL, NULL);
frameLength = WaitForEncodedFrame();
EXPECT_TRUE(frameLength > 0);
EXPECT_TRUE(CheckIfBitExact(_refEncFrame, _refEncFrameLength,
_encodedVideoBuffer.Buffer(), frameLength) == true);
//----- Decoder parameter tests -----
//-- Calls before InitDecode() --
// We want to revert the initialization done in Setup().
EXPECT_TRUE(_decoder->Release() == WEBRTC_VIDEO_CODEC_OK);
VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
EXPECT_TRUE(_decoder->Decode(encodedImage, false, NULL) ==
WEBRTC_VIDEO_CODEC_UNINITIALIZED);
WaitForDecodedFrame();
EXPECT_TRUE(_decoder->Reset() == WEBRTC_VIDEO_CODEC_UNINITIALIZED);
EXPECT_TRUE(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
ASSERT_FALSE(SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK);
//-- Decode() errors --
// Unallocated encodedVideoBuffer.
_encodedVideoBuffer.Free();
VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
encodedImage._length = 10; // Buffer NULL but length > 0
EXPECT_EQ(_decoder->Decode(encodedImage, false, NULL),
WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
_encodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
//----- Decoder stress tests -----
unsigned char* tmpBuf = new unsigned char[_lengthSourceFrame];
// "Random" and zero data.
// We either expect an error, or at the least, no output.
// This relies on the codec's ability to detect an erroneous bitstream.
EXPECT_TRUE(_decoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
EXPECT_TRUE(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
ASSERT_FALSE(SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK);
for (int i = 0; i < 100; i++)
{
ASSERT_TRUE(fread(tmpBuf, 1, _refEncFrameLength, _sourceFile)
== _refEncFrameLength);
_encodedVideoBuffer.CopyFrame(_refEncFrameLength, tmpBuf);
VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
int ret = _decoder->Decode(encodedImage, false, NULL);
EXPECT_TRUE(ret <= 0);
if (ret == 0)
{
EXPECT_TRUE(WaitForDecodedFrame() == 0);
}
memset(tmpBuf, 0, _refEncFrameLength);
_encodedVideoBuffer.CopyFrame(_refEncFrameLength, tmpBuf);
VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
ret = _decoder->Decode(encodedImage, false, NULL);
EXPECT_TRUE(ret <= 0);
if (ret == 0)
{
EXPECT_TRUE(WaitForDecodedFrame() == 0);
}
}
rewind(_sourceFile);
_encodedVideoBuffer.SetLength(_refEncFrameLength);
_encodedVideoBuffer.CopyFrame(_refEncFrameLength, _refEncFrame);
// Init then decode.
EXPECT_TRUE(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
ASSERT_FALSE(SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK);
frameLength = 0;
VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
while (frameLength == 0)
{
_decoder->Decode(encodedImage, false, NULL);
frameLength = WaitForDecodedFrame();
}
EXPECT_TRUE(CheckIfBitExact(_decodedVideoBuffer.Buffer(), frameLength,
_refDecFrame, _lengthSourceFrame) == true);
// Reset then decode.
EXPECT_TRUE(_decoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
frameLength = 0;
VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
while (frameLength == 0)
{
_decoder->Decode(encodedImage, false, NULL);
frameLength = WaitForDecodedFrame();
}
EXPECT_TRUE(CheckIfBitExact(_decodedVideoBuffer.Buffer(), frameLength,
_refDecFrame, _lengthSourceFrame) == true);
// Decode with other size, reset, then decode with original size again
// to verify that decoder is reset to a "fresh" state upon Reset().
{
// Assert that input frame size is a factor of two, so that we can use
// quarter size below.
EXPECT_TRUE((_inst.width % 2 == 0) && (_inst.height % 2 == 0));
VideoCodec tempInst;
memcpy(&tempInst, &_inst, sizeof(VideoCodec));
tempInst.width /= 2;
tempInst.height /= 2;
// Encode reduced (quarter) frame size.
EXPECT_TRUE(_encoder->Release() == WEBRTC_VIDEO_CODEC_OK);
EXPECT_TRUE(_encoder->InitEncode(&tempInst, 1, 1440) ==
WEBRTC_VIDEO_CODEC_OK);
VideoFrame tempInput;
unsigned int tmpLength = _inputVideoBuffer.Length() / 4;
tempInput.CopyFrame(tmpLength, _inputVideoBuffer.Buffer());
tempInput.SetWidth(tempInst.width);
tempInput.SetHeight(tempInst.height);
_encoder->Encode(tempInput, NULL, NULL);
frameLength = WaitForEncodedFrame();
EXPECT_TRUE(frameLength > 0);
tempInput.Free();
// Reset then decode.
EXPECT_TRUE(_decoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
frameLength = 0;
VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
while (frameLength == 0)
{
_decoder->Decode(encodedImage, false, NULL);
frameLength = WaitForDecodedFrame();
}
// Encode original frame again
EXPECT_TRUE(_encoder->Release() == WEBRTC_VIDEO_CODEC_OK);
EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) ==
WEBRTC_VIDEO_CODEC_OK);
_encoder->Encode(_inputVideoBuffer, NULL, NULL);
frameLength = WaitForEncodedFrame();
EXPECT_TRUE(frameLength > 0);
// Reset then decode original frame again.
EXPECT_TRUE(_decoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
frameLength = 0;
VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
while (frameLength == 0)
{
_decoder->Decode(encodedImage, false, NULL);
frameLength = WaitForDecodedFrame();
}
// check that decoded frame matches with reference
EXPECT_TRUE(CheckIfBitExact(_decodedVideoBuffer.Buffer(), frameLength,
_refDecFrame, _lengthSourceFrame) == true);
}
// Release then decode.
EXPECT_TRUE(_decoder->Release() == WEBRTC_VIDEO_CODEC_OK);
EXPECT_TRUE(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
ASSERT_FALSE(SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK);
frameLength = 0;
VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
while (frameLength == 0)
{
_decoder->Decode(encodedImage, false, NULL);
frameLength = WaitForDecodedFrame();
}
EXPECT_TRUE(CheckIfBitExact(_decodedVideoBuffer.Buffer(), frameLength,
_refDecFrame, _lengthSourceFrame) == true);
_encodedVideoBuffer.SetLength(0);
delete [] tmpBuf;
//----- Function tests -----
int frames = 0;
// Do not specify maxBitRate (as in ViE).
_inst.maxBitrate = 0;
//-- Timestamp propagation --
EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
EXPECT_TRUE(_decoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
EXPECT_TRUE(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
ASSERT_FALSE(SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK);
frames = 0;
int frameDelay = 0;
int encTimeStamp;
_decodedVideoBuffer.SetTimeStamp(0);
while (fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile) ==
_lengthSourceFrame)
{
_inputVideoBuffer.CopyFrame(_lengthSourceFrame, _sourceBuffer);
_inputVideoBuffer.SetTimeStamp(frames);
ASSERT_TRUE(_encoder->Encode(_inputVideoBuffer, NULL, NULL) ==
WEBRTC_VIDEO_CODEC_OK);
frameLength = WaitForEncodedFrame();
//ASSERT_TRUE(frameLength);
EXPECT_TRUE(frameLength > 0);
encTimeStamp = _encodedVideoBuffer.TimeStamp();
EXPECT_TRUE(_inputVideoBuffer.TimeStamp() ==
static_cast<unsigned>(encTimeStamp));
frameLength = Decode();
if (frameLength == 0)
{
frameDelay++;
}
encTimeStamp -= frameDelay;
if (encTimeStamp < 0)
{
encTimeStamp = 0;
}
EXPECT_TRUE(_decodedVideoBuffer.TimeStamp() ==
static_cast<unsigned>(encTimeStamp));
frames++;
}
ASSERT_TRUE(feof(_sourceFile) != 0);
rewind(_sourceFile);
RateControlTests();
inputImage.Free();
Teardown();
}
void
UnitTest::RateControlTests()
{
int frames = 0;
VideoFrame inputImage;
WebRtc_UWord32 frameLength;
// Do not specify maxBitRate (as in ViE).
_inst.maxBitrate = 0;
//-- Verify rate control --
EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
EXPECT_TRUE(_decoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
EXPECT_TRUE(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
// add: should also be 0, and 1
const int bitRate[] = {30, 100, 500, 1000, 2000};
const int nBitrates = sizeof(bitRate)/sizeof(*bitRate);
printf("\nRate control test\n");
for (int i = 0; i < nBitrates; i++)
{
_bitRate = bitRate[i];
int totalBytes = 0;
_inst.startBitrate = _bitRate;
_encoder->InitEncode(&_inst, 4, 1440);
_decoder->Reset();
_decoder->InitDecode(&_inst, 1);
frames = 0;
if (_bitRate > _inst.maxBitrate)
{
CodecSpecific_SetBitrate(_bitRate, _inst.maxFramerate);
}
else
{
CodecSpecific_SetBitrate(_bitRate, _inst.maxFramerate);
}
while (fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile) ==
_lengthSourceFrame)
{
_inputVideoBuffer.CopyFrame(_lengthSourceFrame, _sourceBuffer);
_inputVideoBuffer.SetTimeStamp(_inputVideoBuffer.TimeStamp() +
static_cast<WebRtc_UWord32>(9e4 /
static_cast<float>(_inst.maxFramerate)));
ASSERT_EQ(_encoder->Encode(_inputVideoBuffer, NULL, NULL),
WEBRTC_VIDEO_CODEC_OK);
frameLength = WaitForEncodedFrame();
ASSERT_GE(frameLength, 0u);
totalBytes += frameLength;
frames++;
_encodedVideoBuffer.SetLength(0);
}
WebRtc_UWord32 actualBitrate =
(totalBytes / frames * _inst.maxFramerate * 8)/1000;
printf("Target bitrate: %d kbps, actual bitrate: %d kbps\n", _bitRate,
actualBitrate);
// Test for close match over reasonable range.
if (_bitRate >= 100 && _bitRate <= 2500)
{
EXPECT_TRUE(abs(WebRtc_Word32(actualBitrate - _bitRate)) <
0.12 * _bitRate); // for VP8
}
ASSERT_TRUE(feof(_sourceFile) != 0);
rewind(_sourceFile);
}
}
bool
UnitTest::CheckIfBitExact(const void* ptrA, unsigned int aLengthBytes,
const void* ptrB, unsigned int bLengthBytes)
{
if (aLengthBytes != bLengthBytes)
{
return false;
}
return memcmp(ptrA, ptrB, aLengthBytes) == 0;
}

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_UNIT_TEST_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_UNIT_TEST_H_
#include "test.h"
#include "event_wrapper.h"
// Disable "conditional expression is constant" warnings on the perfectly
// acceptable
// do { ... } while (0) constructions below.
// Refer to http://stackoverflow.com/questions/1946445/
// is-there-better-way-to-write-do-while0-construct-to-avoid-compiler-warnings
// for some discussion of the issue.
#ifdef _WIN32
#pragma warning(disable : 4127)
#endif
class VideoSource;
class UnitTestEncodeCompleteCallback;
class UnitTestDecodeCompleteCallback;
class UnitTest : public CodecTest
{
public:
UnitTest();
virtual ~UnitTest();
virtual void Perform();
virtual void Print();
protected:
UnitTest(std::string name, std::string description);
virtual WebRtc_UWord32 CodecSpecific_SetBitrate(
WebRtc_UWord32 bitRate,
WebRtc_UWord32 /* frameRate */);
virtual void Setup();
virtual void Teardown();
virtual void RateControlTests();
virtual int Decode();
virtual int DecodeWithoutAssert();
virtual int SetCodecSpecificParameters() {return 0;};
virtual bool CheckIfBitExact(const void *ptrA, unsigned int aLengthBytes,
const void *ptrB, unsigned int bLengthBytes);
WebRtc_UWord32 WaitForEncodedFrame() const;
WebRtc_UWord32 WaitForDecodedFrame() const;
int _tests;
int _errors;
VideoSource* _source;
unsigned char* _refFrame;
unsigned char* _refEncFrame;
unsigned char* _refDecFrame;
unsigned int _refEncFrameLength;
FILE* _sourceFile;
UnitTestEncodeCompleteCallback* _encodeCompleteCallback;
UnitTestDecodeCompleteCallback* _decodeCompleteCallback;
enum { kMaxWaitEncTimeMs = 100 };
enum { kMaxWaitDecTimeMs = 25 };
};
class UnitTestEncodeCompleteCallback : public webrtc::EncodedImageCallback
{
public:
UnitTestEncodeCompleteCallback(webrtc::VideoFrame* buffer,
WebRtc_UWord32 decoderSpecificSize = 0,
void* decoderSpecificInfo = NULL) :
_encodedVideoBuffer(buffer),
_encodeComplete(false) {}
WebRtc_Word32 Encoded(webrtc::EncodedImage& encodedImage,
const webrtc::CodecSpecificInfo* codecSpecificInfo,
const webrtc::RTPFragmentationHeader*
fragmentation = NULL);
bool EncodeComplete();
// Note that this only makes sense if an encode has been completed
webrtc::VideoFrameType EncodedFrameType() const;
private:
webrtc::VideoFrame* _encodedVideoBuffer;
bool _encodeComplete;
webrtc::VideoFrameType _encodedFrameType;
};
class UnitTestDecodeCompleteCallback : public webrtc::DecodedImageCallback
{
public:
UnitTestDecodeCompleteCallback(webrtc::VideoFrame* buffer) :
_decodedVideoBuffer(buffer), _decodeComplete(false) {}
WebRtc_Word32 Decoded(webrtc::VideoFrame& image);
bool DecodeComplete();
private:
webrtc::VideoFrame* _decodedVideoBuffer;
bool _decodeComplete;
};
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_UNIT_TEST_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "video_source.h"
#include <stdio.h>
#include "gtest/gtest.h"
#include "testsupport/fileutils.h"
VideoSource::VideoSource()
:
_fileName(webrtc::test::ProjectRootPath() + "resources/foreman_cif.yuv"),
_width(352),
_height(288),
_type(webrtc::kI420),
_frameRate(30)
{
}
VideoSource::VideoSource(std::string fileName, VideoSize size,
int frameRate /*= 30*/, webrtc::VideoType type /*= webrtc::kI420*/)
:
_fileName(fileName),
_type(type),
_frameRate(frameRate)
{
assert(size != kUndefined && size != kNumberOfVideoSizes);
assert(type != webrtc::kUnknown);
assert(frameRate > 0);
if (GetWidthHeight(size, _width, _height) != 0) {
assert(false);
}
}
VideoSource::VideoSource(std::string fileName, int width, int height,
int frameRate /*= 30*/, webrtc::VideoType type /*= webrtc::kI420*/)
:
_fileName(fileName),
_width(width),
_height(height),
_type(type),
_frameRate(frameRate)
{
assert(width > 0);
assert(height > 0);
assert(type != webrtc::kUnknown);
assert(frameRate > 0);
}
VideoSize
VideoSource::GetSize() const
{
return GetSize(_width, _height);
}
VideoSize
VideoSource::GetSize(WebRtc_UWord16 width, WebRtc_UWord16 height)
{
if(width == 128 && height == 96)
{
return kSQCIF;
}else if(width == 160 && height == 120)
{
return kQQVGA;
}else if(width == 176 && height == 144)
{
return kQCIF;
}else if(width == 320 && height == 240)
{
return kQVGA;
}else if(width == 352 && height == 288)
{
return kCIF;
}else if(width == 640 && height == 480)
{
return kVGA;
}else if(width == 720 && height == 480)
{
return kNTSC;
}else if(width == 704 && height == 576)
{
return k4CIF;
}else if(width == 800 && height == 600)
{
return kSVGA;
}else if(width == 960 && height == 720)
{
return kHD;
}else if(width == 1024 && height == 768)
{
return kXGA;
}else if(width == 1440 && height == 1080)
{
return kFullHD;
}else if(width == 400 && height == 240)
{
return kWQVGA;
}else if(width == 800 && height == 480)
{
return kWVGA;
}else if(width == 1280 && height == 720)
{
return kWHD;
}else if(width == 1920 && height == 1080)
{
return kWFullHD;
}
return kUndefined;
}
unsigned int
VideoSource::GetFrameLength() const
{
return webrtc::CalcBufferSize(_type, _width, _height);
}
const char*
VideoSource::GetMySizeString() const
{
return VideoSource::GetSizeString(GetSize());
}
const char*
VideoSource::GetSizeString(VideoSize size)
{
switch (size)
{
case kSQCIF:
return "SQCIF";
case kQQVGA:
return "QQVGA";
case kQCIF:
return "QCIF";
case kQVGA:
return "QVGA";
case kCIF:
return "CIF";
case kVGA:
return "VGA";
case kNTSC:
return "NTSC";
case k4CIF:
return "4CIF";
case kSVGA:
return "SVGA";
case kHD:
return "HD";
case kXGA:
return "XGA";
case kFullHD:
return "Full_HD";
case kWQVGA:
return "WQVGA";
case kWHD:
return "WHD";
case kWFullHD:
return "WFull_HD";
default:
return "Undefined";
}
}
std::string
VideoSource::GetFilePath() const
{
size_t slashPos = _fileName.find_last_of("/\\");
if (slashPos == std::string::npos)
{
return ".";
}
return _fileName.substr(0, slashPos);
}
std::string
VideoSource::GetName() const
{
// Remove path.
size_t slashPos = _fileName.find_last_of("/\\");
if (slashPos == std::string::npos)
{
slashPos = 0;
}
else
{
slashPos++;
}
// Remove extension and underscored suffix if it exists.
return _fileName.substr(slashPos, std::min(_fileName.find_last_of("_"),
_fileName.find_last_of(".")) - slashPos);
}
void
VideoSource::Convert(const VideoSource &target, bool force /* = false */) const
{
// Ensure target rate is less than or equal to source
// (i.e. we are only temporally downsampling).
ASSERT_TRUE(target.GetFrameRate() <= _frameRate);
// Only supports YUV420 currently.
ASSERT_TRUE(_type == webrtc::kI420 && target.GetType() == webrtc::kI420);
if (!force && (FileExists(target.GetFileName().c_str()) ||
(target.GetWidth() == _width && target.GetHeight() == _height && target.GetFrameRate() == _frameRate)))
{
// Assume that the filename uniquely defines the content.
// If the file already exists, it is the correct file.
return;
}
FILE *inFile = NULL;
FILE *outFile = NULL;
inFile = fopen(_fileName.c_str(), "rb");
ASSERT_TRUE(inFile != NULL);
outFile = fopen(target.GetFileName().c_str(), "wb");
ASSERT_TRUE(outFile != NULL);
FrameDropper fd;
fd.SetFrameRate(target.GetFrameRate(), _frameRate);
const size_t lengthOutFrame = webrtc::CalcBufferSize(target.GetType(),
target.GetWidth(), target.GetHeight());
ASSERT_TRUE(lengthOutFrame > 0);
unsigned char *outFrame = new unsigned char[lengthOutFrame];
const size_t lengthInFrame = webrtc::CalcBufferSize(_type, _width, _height);
ASSERT_TRUE(lengthInFrame > 0);
unsigned char *inFrame = new unsigned char[lengthInFrame];
while (fread(inFrame, 1, lengthInFrame, inFile) == lengthInFrame)
{
if (!fd.DropFrame())
{
ASSERT_TRUE(target.GetWidth() == _width &&
target.GetHeight() == _height);
// Add video interpolator here!
if (fwrite(outFrame, 1, lengthOutFrame,
outFile) != lengthOutFrame) {
return;
}
}
}
delete inFrame;
delete outFrame;
fclose(inFile);
fclose(outFile);
}
bool VideoSource::FileExists(const char* fileName)
{
FILE* fp = NULL;
fp = fopen(fileName, "rb");
if(fp != NULL)
{
fclose(fp);
return true;
}
return false;
}
int
VideoSource::GetWidthHeight( VideoSize size, int & width, int& height)
{
switch(size)
{
case kSQCIF:
width = 128;
height = 96;
return 0;
case kQQVGA:
width = 160;
height = 120;
return 0;
case kQCIF:
width = 176;
height = 144;
return 0;
case kCGA:
width = 320;
height = 200;
return 0;
case kQVGA:
width = 320;
height = 240;
return 0;
case kSIF:
width = 352;
height = 240;
return 0;
case kWQVGA:
width = 400;
height = 240;
return 0;
case kCIF:
width = 352;
height = 288;
return 0;
case kW288p:
width = 512;
height = 288;
return 0;
case k448p:
width = 576;
height = 448;
return 0;
case kVGA:
width = 640;
height = 480;
return 0;
case k432p:
width = 720;
height = 432;
return 0;
case kW432p:
width = 768;
height = 432;
return 0;
case k4SIF:
width = 704;
height = 480;
return 0;
case kW448p:
width = 768;
height = 448;
return 0;
case kNTSC:
width = 720;
height = 480;
return 0;
case kFW448p:
width = 800;
height = 448;
return 0;
case kWVGA:
width = 800;
height = 480;
return 0;
case k4CIF:
width = 704;
height = 576;
return 0;
case kSVGA:
width = 800;
height = 600;
return 0;
case kW544p:
width = 960;
height = 544;
return 0;
case kW576p:
width = 1024;
height = 576;
return 0;
case kHD:
width = 960;
height = 720;
return 0;
case kXGA:
width = 1024;
height = 768;
return 0;
case kFullHD:
width = 1440;
height = 1080;
return 0;
case kWHD:
width = 1280;
height = 720;
return 0;
case kWFullHD:
width = 1920;
height = 1080;
return 0;
default:
return -1;
}
}
FrameDropper::FrameDropper()
:
_dropsBetweenRenders(0),
_frameCounter(0)
{
}
bool
FrameDropper::DropFrame()
{
_frameCounter++;
if (_frameCounter > _dropsBetweenRenders)
{
_frameCounter = 0;
return false;
}
return true;
}
unsigned int
FrameDropper::DropsBetweenRenders()
{
return _dropsBetweenRenders;
}
void
FrameDropper::SetFrameRate(double frameRate, double maxFrameRate)
{
if (frameRate >= 1.0)
{
_dropsBetweenRenders = static_cast<unsigned int>(maxFrameRate / frameRate + 0.5) - 1;
}
else
{
_dropsBetweenRenders = 0;
}
}

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_VIDEO_SOURCE_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_VIDEO_SOURCE_H_
#include <string>
#include "common_video/libyuv/include/webrtc_libyuv.h"
enum VideoSize
{
kUndefined,
kSQCIF, // 128*96 = 12 288
kQQVGA, // 160*120 = 19 200
kQCIF, // 176*144 = 25 344
kCGA, // 320*200 = 64 000
kQVGA, // 320*240 = 76 800
kSIF, // 352*240 = 84 480
kWQVGA, // 400*240 = 96 000
kCIF, // 352*288 = 101 376
kW288p, // 512*288 = 147 456 (WCIF)
k448p, // 576*448 = 281 088
kVGA, // 640*480 = 307 200
k432p, // 720*432 = 311 040
kW432p, // 768*432 = 331 776
k4SIF, // 704*480 = 337 920
kW448p, // 768*448 = 344 064
kNTSC, // 720*480 = 345 600
kFW448p, // 800*448 = 358 400
kWVGA, // 800*480 = 384 000
k4CIF, // 704<30>576 = 405 504
kSVGA, // 800*600 = 480 000
kW544p, // 960*544 = 522 240
kW576p, // 1024*576 = 589 824 (W4CIF)
kHD, // 960*720 = 691 200
kXGA, // 1024*768 = 786 432
kWHD, // 1280*720 = 921 600
kFullHD, // 1440*1080 = 1 555 200
kWFullHD, // 1920*1080 = 2 073 600
kNumberOfVideoSizes
};
class VideoSource
{
public:
VideoSource();
VideoSource(std::string fileName, VideoSize size, int frameRate = 30,
webrtc::VideoType type = webrtc::kI420);
VideoSource(std::string fileName, int width, int height, int frameRate = 30,
webrtc::VideoType type = webrtc::kI420);
std::string GetFileName() const { return _fileName; }
int GetWidth() const { return _width; }
int GetHeight() const { return _height; }
webrtc::VideoType GetType() const { return _type; }
int GetFrameRate() const { return _frameRate; }
// Returns the file path without a trailing slash.
std::string GetFilePath() const;
// Returns the filename with the path (including the leading slash) removed.
std::string GetName() const;
VideoSize GetSize() const;
static VideoSize GetSize(WebRtc_UWord16 width, WebRtc_UWord16 height);
unsigned int GetFrameLength() const;
// Returns a human-readable size string.
static const char* GetSizeString(VideoSize size);
const char* GetMySizeString() const;
// Opens the video source, converting and writing to the specified target.
// If force is true, the conversion will be done even if the target file
// already exists.
void Convert(const VideoSource& target, bool force = false) const;
static bool FileExists(const char* fileName);
private:
static int GetWidthHeight( VideoSize size, int& width, int& height);
std::string _fileName;
int _width;
int _height;
webrtc::VideoType _type;
int _frameRate;
};
class FrameDropper
{
public:
FrameDropper();
bool DropFrame();
unsigned int DropsBetweenRenders();
void SetFrameRate(double frameRate, double maxFrameRate);
private:
unsigned int _dropsBetweenRenders;
unsigned int _frameCounter;
};
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_VIDEO_SOURCE_H_

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# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'conditions': [
['include_tests==1', {
'targets': [
{
'target_name': 'video_quality_measurement',
'type': 'executable',
'dependencies': [
'video_codecs_test_framework',
'webrtc_video_coding',
'<(DEPTH)/third_party/google-gflags/google-gflags.gyp:google-gflags',
'<(webrtc_root)/test/metrics.gyp:metrics',
'<(webrtc_vp8_dir)/vp8.gyp:webrtc_vp8',
],
'sources': [
'video_quality_measurement.cc',
],
},
], # targets
}], # include_tests
], # conditions
}

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <stdarg.h>
#include <sys/stat.h> // To check for directory existence.
#include <cassert>
#include <cstdio>
#include <ctime>
#ifndef S_ISDIR // Not defined in stat.h on Windows.
#define S_ISDIR(mode) (((mode) & S_IFMT) == S_IFDIR)
#endif
#include "common_types.h"
#include "google/gflags.h"
#include "modules/video_coding/codecs/test/packet_manipulator.h"
#include "modules/video_coding/codecs/test/stats.h"
#include "modules/video_coding/codecs/test/videoprocessor.h"
#include "modules/video_coding/codecs/vp8/include/vp8.h"
#include "modules/video_coding/main/interface/video_coding.h"
#include "system_wrappers/interface/trace.h"
#include "testsupport/frame_reader.h"
#include "testsupport/frame_writer.h"
#include "testsupport/metrics/video_metrics.h"
#include "testsupport/packet_reader.h"
DEFINE_string(test_name, "Quality test", "The name of the test to run. ");
DEFINE_string(test_description, "", "A more detailed description about what "
"the current test is about.");
DEFINE_string(input_filename, "", "Input file. "
"The source video file to be encoded and decoded. Must be in "
".yuv format");
DEFINE_int32(width, -1, "Width in pixels of the frames in the input file.");
DEFINE_int32(height, -1, "Height in pixels of the frames in the input file.");
DEFINE_int32(framerate, 30, "Frame rate of the input file, in FPS "
"(frames-per-second). ");
DEFINE_string(output_dir, ".", "Output directory. "
"The directory where the output file will be put. Must already "
"exist.");
DEFINE_bool(use_single_core, false, "Force using a single core. If set to "
"true, only one core will be used for processing. Using a single "
"core is necessary to get a deterministic behavior for the"
"encoded frames - using multiple cores will produce different "
"encoded frames since multiple cores are competing to consume the "
"byte budget for each frame in parallel. If set to false, "
"the maximum detected number of cores will be used. ");
DEFINE_bool(disable_fixed_random_seed , false, "Set this flag to disable the"
"usage of a fixed random seed for the random generator used "
"for packet loss. Disabling this will cause consecutive runs "
"loose packets at different locations, which is bad for "
"reproducibility.");
DEFINE_string(output_filename, "", "Output file. "
"The name of the output video file resulting of the processing "
"of the source file. By default this is the same name as the "
"input file with '_out' appended before the extension.");
DEFINE_int32(bitrate, 500, "Bit rate in kilobits/second.");
DEFINE_int32(keyframe_interval, 0, "Forces a keyframe every Nth frame. "
"0 means the encoder decides when to insert keyframes. Note that "
"the encoder may create a keyframe in other locations in addition "
"to the interval that is set using this parameter.");
DEFINE_int32(temporal_layers, 0, "The number of temporal layers to use "
"(VP8 specific codec setting). Must be 0-4.");
DEFINE_int32(packet_size, 1500, "Simulated network packet size in bytes (MTU). "
"Used for packet loss simulation.");
DEFINE_int32(max_payload_size, 1440, "Max payload size in bytes for the "
"encoder.");
DEFINE_string(packet_loss_mode, "uniform", "Packet loss mode. Two different "
"packet loss models are supported: uniform or burst. This "
"setting has no effect unless packet_loss_rate is >0. ");
DEFINE_double(packet_loss_probability, 0.0, "Packet loss probability. A value "
"between 0.0 and 1.0 that defines the probability of a packet "
"being lost. 0.1 means 10% and so on.");
DEFINE_int32(packet_loss_burst_length, 1, "Packet loss burst length. Defines "
"how many packets will be lost in a burst when a packet has been "
"decided to be lost. Must be >=1.");
DEFINE_bool(csv, false, "CSV output. Enabling this will output all frame "
"statistics at the end of execution. Recommended to run combined "
"with --noverbose to avoid mixing output.");
DEFINE_bool(python, false, "Python output. Enabling this will output all frame "
"statistics as a Python script at the end of execution. "
"Recommended to run combine with --noverbose to avoid mixing "
"output.");
DEFINE_bool(verbose, true, "Verbose mode. Prints a lot of debugging info. "
"Suitable for tracking progress but not for capturing output. "
"Disable with --noverbose flag.");
// Custom log method that only prints if the verbose flag is given.
// Supports all the standard printf parameters and formatting (just forwarded).
int Log(const char *format, ...) {
int result = 0;
if (FLAGS_verbose) {
va_list args;
va_start(args, format);
result = vprintf(format, args);
va_end(args);
}
return result;
}
// Validates the arguments given as command line flags and fills in the
// TestConfig struct with all configurations needed for video processing.
// Returns 0 if everything is OK, otherwise an exit code.
int HandleCommandLineFlags(webrtc::test::TestConfig* config) {
// Validate the mandatory flags:
if (FLAGS_input_filename == "" || FLAGS_width == -1 || FLAGS_height == -1) {
printf("%s\n", google::ProgramUsage());
return 1;
}
config->name = FLAGS_test_name;
config->description = FLAGS_test_description;
// Verify the input file exists and is readable.
FILE* test_file;
test_file = fopen(FLAGS_input_filename.c_str(), "rb");
if (test_file == NULL) {
fprintf(stderr, "Cannot read the specified input file: %s\n",
FLAGS_input_filename.c_str());
return 2;
}
fclose(test_file);
config->input_filename = FLAGS_input_filename;
// Verify the output dir exists.
struct stat dir_info;
if (!(stat(FLAGS_output_dir.c_str(), &dir_info) == 0 &&
S_ISDIR(dir_info.st_mode))) {
fprintf(stderr, "Cannot find output directory: %s\n",
FLAGS_output_dir.c_str());
return 3;
}
config->output_dir = FLAGS_output_dir;
// Manufacture an output filename if none was given.
if (FLAGS_output_filename == "") {
// Cut out the filename without extension from the given input file
// (which may include a path)
int startIndex = FLAGS_input_filename.find_last_of("/") + 1;
if (startIndex == 0) {
startIndex = 0;
}
FLAGS_output_filename =
FLAGS_input_filename.substr(startIndex,
FLAGS_input_filename.find_last_of(".")
- startIndex) + "_out.yuv";
}
// Verify output file can be written.
if (FLAGS_output_dir == ".") {
config->output_filename = FLAGS_output_filename;
} else {
config->output_filename = FLAGS_output_dir + "/"+ FLAGS_output_filename;
}
test_file = fopen(config->output_filename.c_str(), "wb");
if (test_file == NULL) {
fprintf(stderr, "Cannot write output file: %s\n",
config->output_filename.c_str());
return 4;
}
fclose(test_file);
// Check single core flag.
config->use_single_core = FLAGS_use_single_core;
// Get codec specific configuration.
webrtc::VideoCodingModule::Codec(webrtc::kVideoCodecVP8,
config->codec_settings);
// Check the temporal layers.
if (FLAGS_temporal_layers < 0 ||
FLAGS_temporal_layers > webrtc::kMaxTemporalStreams) {
fprintf(stderr, "Temporal layers number must be 0-4, was: %d\n",
FLAGS_temporal_layers);
return 13;
}
config->codec_settings->codecSpecific.VP8.numberOfTemporalLayers =
FLAGS_temporal_layers;
// Check the bit rate.
if (FLAGS_bitrate <= 0) {
fprintf(stderr, "Bit rate must be >0 kbps, was: %d\n", FLAGS_bitrate);
return 5;
}
config->codec_settings->startBitrate = FLAGS_bitrate;
// Check the keyframe interval.
if (FLAGS_keyframe_interval < 0) {
fprintf(stderr, "Keyframe interval must be >=0, was: %d\n",
FLAGS_keyframe_interval);
return 6;
}
config->keyframe_interval = FLAGS_keyframe_interval;
// Check packet size and max payload size.
if (FLAGS_packet_size <= 0) {
fprintf(stderr, "Packet size must be >0 bytes, was: %d\n",
FLAGS_packet_size);
return 7;
}
config->networking_config.packet_size_in_bytes = FLAGS_packet_size;
if (FLAGS_max_payload_size <= 0) {
fprintf(stderr, "Max payload size must be >0 bytes, was: %d\n",
FLAGS_max_payload_size);
return 8;
}
config->networking_config.max_payload_size_in_bytes =
FLAGS_max_payload_size;
// Check the width and height
if (FLAGS_width <= 0 || FLAGS_height <= 0) {
fprintf(stderr, "Width and height must be >0.");
return 9;
}
config->codec_settings->width = FLAGS_width;
config->codec_settings->height = FLAGS_height;
config->codec_settings->maxFramerate = FLAGS_framerate;
// Calculate the size of each frame to read (according to YUV spec).
config->frame_length_in_bytes =
3 * config->codec_settings->width * config->codec_settings->height / 2;
// Check packet loss settings
if (FLAGS_packet_loss_mode != "uniform" &&
FLAGS_packet_loss_mode != "burst") {
fprintf(stderr, "Unsupported packet loss mode, must be 'uniform' or "
"'burst'\n.");
return 10;
}
config->networking_config.packet_loss_mode = webrtc::test::kUniform;
if (FLAGS_packet_loss_mode == "burst") {
config->networking_config.packet_loss_mode = webrtc::test::kBurst;
}
if (FLAGS_packet_loss_probability < 0.0 ||
FLAGS_packet_loss_probability > 1.0) {
fprintf(stderr, "Invalid packet loss probability. Must be 0.0 - 1.0, "
"was: %f\n", FLAGS_packet_loss_probability);
return 11;
}
config->networking_config.packet_loss_probability =
FLAGS_packet_loss_probability;
if (FLAGS_packet_loss_burst_length < 1) {
fprintf(stderr, "Invalid packet loss burst length, must be >=1, "
"was: %d\n", FLAGS_packet_loss_burst_length);
return 12;
}
config->networking_config.packet_loss_burst_length =
FLAGS_packet_loss_burst_length;
config->verbose = FLAGS_verbose;
return 0;
}
void CalculateSsimVideoMetrics(webrtc::test::TestConfig* config,
webrtc::test::QualityMetricsResult* result) {
Log("Calculating SSIM...\n");
I420SSIMFromFiles(config->input_filename.c_str(),
config->output_filename.c_str(),
config->codec_settings->width,
config->codec_settings->height, result);
Log(" Average: %3.2f\n", result->average);
Log(" Min : %3.2f (frame %d)\n", result->min, result->min_frame_number);
Log(" Max : %3.2f (frame %d)\n", result->max, result->max_frame_number);
}
void CalculatePsnrVideoMetrics(webrtc::test::TestConfig* config,
webrtc::test::QualityMetricsResult* result) {
Log("Calculating PSNR...\n");
I420PSNRFromFiles(config->input_filename.c_str(),
config->output_filename.c_str(),
config->codec_settings->width,
config->codec_settings->height, result);
Log(" Average: %3.2f\n", result->average);
Log(" Min : %3.2f (frame %d)\n", result->min, result->min_frame_number);
Log(" Max : %3.2f (frame %d)\n", result->max, result->max_frame_number);
}
void PrintConfigurationSummary(const webrtc::test::TestConfig& config) {
Log("Quality test with parameters:\n");
Log(" Test name : %s\n", config.name.c_str());
Log(" Description : %s\n", config.description.c_str());
Log(" Input filename : %s\n", config.input_filename.c_str());
Log(" Output directory : %s\n", config.output_dir.c_str());
Log(" Output filename : %s\n", config.output_filename.c_str());
Log(" Frame length : %d bytes\n", config.frame_length_in_bytes);
Log(" Packet size : %d bytes\n",
config.networking_config.packet_size_in_bytes);
Log(" Max payload size : %d bytes\n",
config.networking_config.max_payload_size_in_bytes);
Log(" Packet loss:\n");
Log(" Mode : %s\n",
PacketLossModeToStr(config.networking_config.packet_loss_mode));
Log(" Probability : %2.1f\n",
config.networking_config.packet_loss_probability);
Log(" Burst length : %d packets\n",
config.networking_config.packet_loss_burst_length);
}
void PrintCsvOutput(const webrtc::test::Stats& stats,
const webrtc::test::QualityMetricsResult& ssim_result,
const webrtc::test::QualityMetricsResult& psnr_result) {
Log("\nCSV output (recommended to run with --noverbose to skip the "
"above output)\n");
printf("frame_number encoding_successful decoding_successful "
"encode_return_code decode_return_code "
"encode_time_in_us decode_time_in_us "
"bit_rate_in_kbps encoded_frame_length_in_bytes frame_type "
"packets_dropped total_packets "
"ssim psnr\n");
for (unsigned int i = 0; i < stats.stats_.size(); ++i) {
const webrtc::test::FrameStatistic& f = stats.stats_[i];
const webrtc::test::FrameResult& ssim = ssim_result.frames[i];
const webrtc::test::FrameResult& psnr = psnr_result.frames[i];
printf("%4d, %d, %d, %2d, %2d, %6d, %6d, %5d, %7d, %d, %2d, %2d, "
"%5.3f, %5.2f\n",
f.frame_number,
f.encoding_successful,
f.decoding_successful,
f.encode_return_code,
f.decode_return_code,
f.encode_time_in_us,
f.decode_time_in_us,
f.bit_rate_in_kbps,
f.encoded_frame_length_in_bytes,
f.frame_type,
f.packets_dropped,
f.total_packets,
ssim.value,
psnr.value);
}
}
void PrintPythonOutput(const webrtc::test::TestConfig& config,
const webrtc::test::Stats& stats,
const webrtc::test::QualityMetricsResult& ssim_result,
const webrtc::test::QualityMetricsResult& psnr_result) {
Log("\nPython output (recommended to run with --noverbose to skip the "
"above output)\n");
printf("test_configuration = ["
"{'name': 'name', 'value': '%s'},\n"
"{'name': 'description', 'value': '%s'},\n"
"{'name': 'test_number', 'value': '%d'},\n"
"{'name': 'input_filename', 'value': '%s'},\n"
"{'name': 'output_filename', 'value': '%s'},\n"
"{'name': 'output_dir', 'value': '%s'},\n"
"{'name': 'packet_size_in_bytes', 'value': '%d'},\n"
"{'name': 'max_payload_size_in_bytes', 'value': '%d'},\n"
"{'name': 'packet_loss_mode', 'value': '%s'},\n"
"{'name': 'packet_loss_probability', 'value': '%f'},\n"
"{'name': 'packet_loss_burst_length', 'value': '%d'},\n"
"{'name': 'exclude_frame_types', 'value': '%s'},\n"
"{'name': 'frame_length_in_bytes', 'value': '%d'},\n"
"{'name': 'use_single_core', 'value': '%s'},\n"
"{'name': 'keyframe_interval;', 'value': '%d'},\n"
"{'name': 'video_codec_type', 'value': '%s'},\n"
"{'name': 'width', 'value': '%d'},\n"
"{'name': 'height', 'value': '%d'},\n"
"{'name': 'bit_rate_in_kbps', 'value': '%d'},\n"
"]\n",
config.name.c_str(),
config.description.c_str(),
config.test_number,
config.input_filename.c_str(),
config.output_filename.c_str(),
config.output_dir.c_str(),
config.networking_config.packet_size_in_bytes,
config.networking_config.max_payload_size_in_bytes,
PacketLossModeToStr(config.networking_config.packet_loss_mode),
config.networking_config.packet_loss_probability,
config.networking_config.packet_loss_burst_length,
ExcludeFrameTypesToStr(config.exclude_frame_types),
config.frame_length_in_bytes,
config.use_single_core ? "True " : "False",
config.keyframe_interval,
webrtc::test::VideoCodecTypeToStr(config.codec_settings->codecType),
config.codec_settings->width,
config.codec_settings->height,
config.codec_settings->startBitrate);
printf("frame_data_types = {"
"'frame_number': ('number', 'Frame number'),\n"
"'encoding_successful': ('boolean', 'Encoding successful?'),\n"
"'decoding_successful': ('boolean', 'Decoding successful?'),\n"
"'encode_time': ('number', 'Encode time (us)'),\n"
"'decode_time': ('number', 'Decode time (us)'),\n"
"'encode_return_code': ('number', 'Encode return code'),\n"
"'decode_return_code': ('number', 'Decode return code'),\n"
"'bit_rate': ('number', 'Bit rate (kbps)'),\n"
"'encoded_frame_length': "
"('number', 'Encoded frame length (bytes)'),\n"
"'frame_type': ('string', 'Frame type'),\n"
"'packets_dropped': ('number', 'Packets dropped'),\n"
"'total_packets': ('number', 'Total packets'),\n"
"'ssim': ('number', 'SSIM'),\n"
"'psnr': ('number', 'PSNR (dB)'),\n"
"}\n");
printf("frame_data = [");
for (unsigned int i = 0; i < stats.stats_.size(); ++i) {
const webrtc::test::FrameStatistic& f = stats.stats_[i];
const webrtc::test::FrameResult& ssim = ssim_result.frames[i];
const webrtc::test::FrameResult& psnr = psnr_result.frames[i];
printf("{'frame_number': %d, "
"'encoding_successful': %s, 'decoding_successful': %s, "
"'encode_time': %d, 'decode_time': %d, "
"'encode_return_code': %d, 'decode_return_code': %d, "
"'bit_rate': %d, 'encoded_frame_length': %d, 'frame_type': %s, "
"'packets_dropped': %d, 'total_packets': %d, "
"'ssim': %f, 'psnr': %f},\n",
f.frame_number,
f.encoding_successful ? "True " : "False",
f.decoding_successful ? "True " : "False",
f.encode_time_in_us,
f.decode_time_in_us,
f.encode_return_code,
f.decode_return_code,
f.bit_rate_in_kbps,
f.encoded_frame_length_in_bytes,
f.frame_type == webrtc::kDeltaFrame ? "'Delta'" : "'Other'",
f.packets_dropped,
f.total_packets,
ssim.value,
psnr.value);
}
printf("]\n");
}
// Runs a quality measurement on the input file supplied to the program.
// The input file must be in YUV format.
int main(int argc, char* argv[]) {
std::string program_name = argv[0];
std::string usage = "Quality test application for video comparisons.\n"
"Run " + program_name + " --helpshort for usage.\n"
"Example usage:\n" + program_name +
" --input_filename=filename.yuv --width=352 --height=288\n";
google::SetUsageMessage(usage);
google::ParseCommandLineFlags(&argc, &argv, true);
// Create TestConfig and codec settings struct.
webrtc::test::TestConfig config;
webrtc::VideoCodec codec_settings;
config.codec_settings = &codec_settings;
int return_code = HandleCommandLineFlags(&config);
// Exit if an invalid argument is supplied.
if (return_code != 0) {
return return_code;
}
PrintConfigurationSummary(config);
webrtc::VP8Encoder* encoder = webrtc::VP8Encoder::Create();
webrtc::VP8Decoder* decoder = webrtc::VP8Decoder::Create();
webrtc::test::Stats stats;
webrtc::test::FrameReaderImpl frame_reader(config.input_filename,
config.frame_length_in_bytes);
webrtc::test::FrameWriterImpl frame_writer(config.output_filename,
config.frame_length_in_bytes);
frame_reader.Init();
frame_writer.Init();
webrtc::test::PacketReader packet_reader;
webrtc::test::PacketManipulatorImpl packet_manipulator(
&packet_reader, config.networking_config, config.verbose);
// By default the packet manipulator is seeded with a fixed random.
// If disabled we must generate a new seed.
if (FLAGS_disable_fixed_random_seed) {
packet_manipulator.InitializeRandomSeed(time(NULL));
}
webrtc::test::VideoProcessor* processor =
new webrtc::test::VideoProcessorImpl(encoder, decoder,
&frame_reader,
&frame_writer,
&packet_manipulator,
config, &stats);
processor->Init();
int frame_number = 0;
while (processor->ProcessFrame(frame_number)) {
if (frame_number % 80 == 0) {
Log("\n"); // make the output a bit nicer.
}
Log(".");
frame_number++;
}
Log("\n");
Log("Processed %d frames\n", frame_number);
// Release encoder and decoder to make sure they have finished processing.
encoder->Release();
decoder->Release();
// Verify statistics are correct:
assert(frame_number == static_cast<int>(stats.stats_.size()));
// Close the files before we start using them for SSIM/PSNR calculations.
frame_reader.Close();
frame_writer.Close();
stats.PrintSummary();
webrtc::test::QualityMetricsResult ssim_result;
CalculateSsimVideoMetrics(&config, &ssim_result);
webrtc::test::QualityMetricsResult psnr_result;
CalculatePsnrVideoMetrics(&config, &psnr_result);
if (FLAGS_csv) {
PrintCsvOutput(stats, ssim_result, psnr_result);
}
if (FLAGS_python) {
PrintPythonOutput(config, stats, ssim_result, psnr_result);
}
delete processor;
delete encoder;
delete decoder;
Log("Quality test finished!");
return 0;
}

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# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
LOCAL_PATH := $(call my-dir)
include $(CLEAR_VARS)
include $(LOCAL_PATH)/../../../../../../../android-webrtc.mk
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
LOCAL_MODULE := libwebrtc_vp8
LOCAL_MODULE_TAGS := optional
LOCAL_CPP_EXTENSION := .cc
LOCAL_SRC_FILES := \
reference_picture_selection.cc \
vp8_impl.cc
# Flags passed to both C and C++ files.
LOCAL_CFLAGS := \
$(MY_WEBRTC_COMMON_DEFS)
# TODO(leozwang) Enable WEBRTC_LIBVPX_VERSION after libvpx is updateed
# to a new version and also add temporal_layers.cc
LOCAL_C_INCLUDES := \
$(LOCAL_PATH)/../interface \
$(LOCAL_PATH)/../../../interface \
$(LOCAL_PATH)/../../../../../.. \
$(LOCAL_PATH)/../../../../../../common_video/interface \
$(LOCAL_PATH)/../../../../../../common_video/vplib/main/interface \
$(LOCAL_PATH)/../../../../../../modules/interface \
$(LOCAL_PATH)/../../../../../../system_wrappers/interface \
external/libvpx
LOCAL_SHARED_LIBRARIES := \
libcutils \
libdl \
libstlport
ifndef NDK_ROOT
include external/stlport/libstlport.mk
endif
include $(BUILD_STATIC_LIBRARY)

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*
* WEBRTC VP8 wrapper interface
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_
#include "modules/video_coding/codecs/interface/video_codec_interface.h"
namespace webrtc {
class VP8Encoder : public VideoEncoder {
public:
static VP8Encoder* Create();
virtual ~VP8Encoder() {};
}; // end of VP8Encoder class
class VP8Decoder : public VideoDecoder {
public:
static VP8Decoder* Create();
virtual ~VP8Decoder() {};
}; // end of VP8Decoder class
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_COMMON_TYPES_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_COMMON_TYPES_H_
#include "common_types.h"
namespace webrtc {
// Ratio allocation between temporal streams:
// Values as required for the VP8 codec (accumulating).
static const float
kVp8LayerRateAlloction[kMaxTemporalStreams][kMaxTemporalStreams] = {
{1.0f, 0, 0, 0}, // 1 layer
{0.6f, 1.0f , 0 , 0}, // 2 layers {60%, 40%}
{0.4f, 0.6f , 1.0f, 0}, // 3 layers {40%, 20%, 40%}
{0.25f, 0.4f, 0.6f, 1.0f} // 4 layers {25%, 15%, 20%, 40%}
};
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_COMMON_TYPES_H_

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "reference_picture_selection.h"
#include "typedefs.h"
#include "vpx/vpx_encoder.h"
#include "vpx/vp8cx.h"
namespace webrtc {
ReferencePictureSelection::ReferencePictureSelection()
: kRttConfidence(1.33),
update_golden_next_(true),
established_golden_(false),
received_ack_(false),
last_sent_ref_picture_id_(0),
last_sent_ref_update_time_(0),
established_ref_picture_id_(0),
last_refresh_time_(0),
rtt_(0) {
}
void ReferencePictureSelection::Init() {
update_golden_next_ = true;
established_golden_ = false;
received_ack_ = false;
last_sent_ref_picture_id_ = 0;
last_sent_ref_update_time_ = 0;
established_ref_picture_id_ = 0;
last_refresh_time_ = 0;
rtt_ = 0;
}
void ReferencePictureSelection::ReceivedRPSI(int rpsi_picture_id) {
// Assume RPSI is signaled with 14 bits.
if ((rpsi_picture_id & 0x3fff) == (last_sent_ref_picture_id_ & 0x3fff)) {
// Remote peer has received our last reference frame, switch frame type.
received_ack_ = true;
established_golden_ = update_golden_next_;
update_golden_next_ = !update_golden_next_;
established_ref_picture_id_ = last_sent_ref_picture_id_;
}
}
bool ReferencePictureSelection::ReceivedSLI(uint32_t now_ts) {
bool send_refresh = false;
// Don't send a refresh more than once per round-trip time.
// This is to avoid too frequent refreshes, since the receiver
// will signal an SLI for every corrupt frame.
if (TimestampDiff(now_ts, last_refresh_time_) > rtt_) {
send_refresh = true;
last_refresh_time_ = now_ts;
}
return send_refresh;
}
int ReferencePictureSelection::EncodeFlags(int picture_id, bool send_refresh,
uint32_t now_ts) {
int flags = 0;
// We can't refresh the decoder until we have established the key frame.
if (send_refresh && received_ack_) {
flags |= VP8_EFLAG_NO_REF_LAST; // Don't reference the last frame
if (established_golden_)
flags |= VP8_EFLAG_NO_REF_ARF; // Don't reference the alt-ref frame.
else
flags |= VP8_EFLAG_NO_REF_GF; // Don't reference the golden frame
}
// Make sure we don't update the reference frames too often. We must wait long
// enough for an RPSI to arrive after the decoder decoded the reference frame.
// Ideally that should happen after one round-trip time.
// Add a margin defined by |kRttConfidence|.
uint32_t update_interval = kRttConfidence * rtt_;
if (update_interval < kMinUpdateInterval)
update_interval = kMinUpdateInterval;
// Don't send reference frame updates until we have an established reference.
if (TimestampDiff(now_ts, last_sent_ref_update_time_) > update_interval &&
received_ack_) {
flags |= VP8_EFLAG_NO_REF_LAST; // Don't reference the last frame.
if (update_golden_next_) {
flags |= VP8_EFLAG_FORCE_GF; // Update the golden reference.
flags |= VP8_EFLAG_NO_UPD_ARF; // Don't update alt-ref.
flags |= VP8_EFLAG_NO_REF_GF; // Don't reference the golden frame.
} else {
flags |= VP8_EFLAG_FORCE_ARF; // Update the alt-ref reference.
flags |= VP8_EFLAG_NO_UPD_GF; // Don't update the golden frame.
flags |= VP8_EFLAG_NO_REF_ARF; // Don't reference the alt-ref frame.
}
last_sent_ref_picture_id_ = picture_id;
last_sent_ref_update_time_ = now_ts;
} else {
// No update of golden or alt-ref. We can therefore freely reference the
// established reference frame and the last frame.
if (established_golden_)
flags |= VP8_EFLAG_NO_REF_ARF; // Don't reference the alt-ref frame.
else
flags |= VP8_EFLAG_NO_REF_GF; // Don't reference the golden frame.
flags |= VP8_EFLAG_NO_UPD_GF; // Don't update the golden frame.
flags |= VP8_EFLAG_NO_UPD_ARF; // Don't update the alt-ref frame.
}
return flags;
}
void ReferencePictureSelection::EncodedKeyFrame(int picture_id) {
last_sent_ref_picture_id_ = picture_id;
received_ack_ = false;
}
void ReferencePictureSelection::SetRtt(int rtt) {
// Convert from milliseconds to timestamp frequency.
rtt_ = 90 * rtt;
}
uint32_t ReferencePictureSelection::TimestampDiff(uint32_t new_ts,
uint32_t old_ts) {
if (old_ts > new_ts) {
// Assuming this is a wrap, doing a compensated subtraction.
return (new_ts + (static_cast<int64_t>(1) << 32)) - old_ts;
}
return new_ts - old_ts;
}
} // namespace webrtc

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* This file defines classes for doing reference picture selection, primarily
* with VP8.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_REFERENCE_PICTURE_SELECTION_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_REFERENCE_PICTURE_SELECTION_H_
#include "typedefs.h"
namespace webrtc {
class ReferencePictureSelection {
public:
ReferencePictureSelection();
void Init();
// Report a received reference picture selection indication. This will
// introduce a new established reference if the received RPSI isn't too late.
void ReceivedRPSI(int rpsi_picture_id);
// Report a received slice loss indication. Returns true if a refresh frame
// must be sent to the receiver, which is accomplished by only predicting
// from the established reference.
// |now_ts| is the RTP timestamp corresponding to the current time. Typically
// the capture timestamp of the frame currently being processed.
// Returns true if it's time to encode a decoder refresh, otherwise false.
bool ReceivedSLI(uint32_t now_ts);
// Returns the recommended VP8 encode flags needed. May refresh the decoder
// and/or update the reference buffers.
// |picture_id| picture id of the frame to be encoded.
// |send_refresh| should be set to true if a decoder refresh should be
// encoded, otherwise false.
// |now_ts| is the RTP timestamp corresponding to the current time. Typically
// the capture timestamp of the frame currently being processed.
// Returns the flags to be given to the libvpx encoder when encoding the next
// frame.
int EncodeFlags(int picture_id, bool send_refresh, uint32_t now_ts);
// Notify the RPS that the frame with picture id |picture_id| was encoded as
// a key frame, effectively updating all reference buffers.
void EncodedKeyFrame(int picture_id);
// Set the round-trip time between the sender and the receiver to |rtt|
// milliseconds.
void SetRtt(int rtt);
private:
static uint32_t TimestampDiff(uint32_t new_ts, uint32_t old_ts);
// The minimum time between reference frame updates.
enum { kMinUpdateInterval = 90 * 10 }; // Timestamp frequency
const double kRttConfidence;
bool update_golden_next_;
bool established_golden_;
bool received_ack_;
int last_sent_ref_picture_id_;
uint32_t last_sent_ref_update_time_;
int established_ref_picture_id_;
uint32_t last_refresh_time_;
uint32_t rtt_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_REFERENCE_PICTURE_SELECTION_H_

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "gtest/gtest.h"
#include "reference_picture_selection.h"
#include "vpx/vpx_encoder.h"
#include "vpx/vp8cx.h"
using webrtc::ReferencePictureSelection;
// The minimum time between reference frame updates. Should match the values
// set in reference_picture_selection.h
enum { kMinUpdateInterval = 10 };
// The minimum time between decoder refreshes through restricted prediction.
// Should match the values set in reference_picture_selection.h
enum { kRtt = 10 };
enum {
kNoPropagationGolden = VP8_EFLAG_NO_REF_ARF |
VP8_EFLAG_NO_UPD_GF |
VP8_EFLAG_NO_UPD_ARF,
kNoPropagationAltRef = VP8_EFLAG_NO_REF_GF |
VP8_EFLAG_NO_UPD_GF |
VP8_EFLAG_NO_UPD_ARF,
kPropagateGolden = VP8_EFLAG_FORCE_GF |
VP8_EFLAG_NO_UPD_ARF |
VP8_EFLAG_NO_REF_GF |
VP8_EFLAG_NO_REF_LAST,
kPropagateAltRef = VP8_EFLAG_FORCE_ARF |
VP8_EFLAG_NO_UPD_GF |
VP8_EFLAG_NO_REF_ARF |
VP8_EFLAG_NO_REF_LAST,
kRefreshFromGolden = VP8_EFLAG_NO_REF_LAST |
VP8_EFLAG_NO_REF_ARF,
kRefreshFromAltRef = VP8_EFLAG_NO_REF_LAST |
VP8_EFLAG_NO_REF_GF
};
class TestRPS : public ::testing::Test {
protected:
virtual void SetUp() {
rps_.Init();
// Initialize with sending a key frame and acknowledging it.
rps_.EncodedKeyFrame(0);
rps_.ReceivedRPSI(0);
rps_.SetRtt(kRtt);
}
ReferencePictureSelection rps_;
};
TEST_F(TestRPS, TestPropagateReferenceFrames) {
// Should propagate the alt-ref reference.
uint32_t time = (4 * kMinUpdateInterval) / 3 + 1;
EXPECT_EQ(rps_.EncodeFlags(1, false, 90 * time), kPropagateAltRef);
rps_.ReceivedRPSI(1);
time += (4 * (time + kMinUpdateInterval)) / 3 + 1;
// Should propagate the golden reference.
EXPECT_EQ(rps_.EncodeFlags(2, false, 90 * time), kPropagateGolden);
rps_.ReceivedRPSI(2);
// Should propagate the alt-ref reference.
time = (4 * (time + kMinUpdateInterval)) / 3 + 1;
EXPECT_EQ(rps_.EncodeFlags(3, false, 90 * time), kPropagateAltRef);
rps_.ReceivedRPSI(3);
// Shouldn't propagate any reference frames (except last), and the established
// reference is alt-ref.
time = time + kMinUpdateInterval;
EXPECT_EQ(rps_.EncodeFlags(4, false, 90 * time), kNoPropagationAltRef);
}
TEST_F(TestRPS, TestDecoderRefresh) {
uint32_t time = kRtt + 1;
// No more than one refresh per RTT.
EXPECT_EQ(rps_.ReceivedSLI(90 * time), true);
time += 5;
EXPECT_EQ(rps_.ReceivedSLI(90 * time), false);
time += kRtt - 4;
EXPECT_EQ(rps_.ReceivedSLI(90 * time), true);
// Enough time have elapsed since the previous reference propagation, we will
// therefore get both a refresh from golden and a propagation of alt-ref.
EXPECT_EQ(rps_.EncodeFlags(5, true, 90 * time), kRefreshFromGolden |
kPropagateAltRef);
rps_.ReceivedRPSI(5);
time += kRtt + 1;
// Enough time for a new refresh, but not enough time for a reference
// propagation.
EXPECT_EQ(rps_.ReceivedSLI(90 * time), true);
EXPECT_EQ(rps_.EncodeFlags(6, true, 90 * time), kRefreshFromAltRef |
kNoPropagationAltRef);
}
TEST_F(TestRPS, TestWrap) {
EXPECT_EQ(rps_.ReceivedSLI(0xffffffff), true);
EXPECT_EQ(rps_.ReceivedSLI(1), false);
EXPECT_EQ(rps_.ReceivedSLI(90 * 100), true);
EXPECT_EQ(rps_.EncodeFlags(7, false, 0xffffffff), kPropagateAltRef);
EXPECT_EQ(rps_.EncodeFlags(8, false, 1), kNoPropagationGolden);
EXPECT_EQ(rps_.EncodeFlags(10, false, 90 * 100), kPropagateAltRef);
}

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/* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "temporal_layers.h"
#include <stdlib.h>
#include <string.h>
#include <cassert>
#include "modules/interface/module_common_types.h"
#include "modules/video_coding/codecs/interface/video_codec_interface.h"
#include "modules/video_coding/codecs/vp8/include/vp8_common_types.h"
#include "vpx/vpx_encoder.h"
#include "vpx/vp8cx.h"
namespace webrtc {
TemporalLayers::TemporalLayers(int numberOfTemporalLayers)
: number_of_temporal_layers_(numberOfTemporalLayers),
temporal_ids_length_(0),
temporal_pattern_length_(0),
tl0_pic_idx_(rand()),
pattern_idx_(255),
timestamp_(0) {
assert(kMaxTemporalStreams >= numberOfTemporalLayers);
memset(temporal_ids_, 0, sizeof(temporal_ids_));
memset(temporal_pattern_, 0, sizeof(temporal_pattern_));
}
bool TemporalLayers::ConfigureBitrates(int bitrateKbit,
vpx_codec_enc_cfg_t* cfg) {
switch (number_of_temporal_layers_) {
case 0:
case 1:
// Do nothing.
break;
case 2:
temporal_ids_length_ = 2;
temporal_ids_[0] = 0;
temporal_ids_[1] = 1;
cfg->ts_number_layers = number_of_temporal_layers_;
cfg->ts_periodicity = temporal_ids_length_;
// Split stream 60% 40%.
// Bitrate API for VP8 is the agregated bitrate for all lower layers.
cfg->ts_target_bitrate[0] = bitrateKbit * kVp8LayerRateAlloction[1][0];
cfg->ts_target_bitrate[1] = bitrateKbit;
cfg->ts_rate_decimator[0] = 2;
cfg->ts_rate_decimator[1] = 1;
memcpy(cfg->ts_layer_id,
temporal_ids_,
sizeof(unsigned int) * temporal_ids_length_);
temporal_pattern_length_ = 8;
temporal_pattern_[0] = kTemporalUpdateLastAndGoldenRefAltRef;
temporal_pattern_[1] = kTemporalUpdateGoldenWithoutDependencyRefAltRef;
temporal_pattern_[2] = kTemporalUpdateLastRefAltRef;
temporal_pattern_[3] = kTemporalUpdateGoldenRefAltRef;
temporal_pattern_[4] = kTemporalUpdateLastRefAltRef;
temporal_pattern_[5] = kTemporalUpdateGoldenRefAltRef;
temporal_pattern_[6] = kTemporalUpdateLastRefAltRef;
temporal_pattern_[7] = kTemporalUpdateNone;
break;
case 3:
temporal_ids_length_ = 4;
temporal_ids_[0] = 0;
temporal_ids_[1] = 2;
temporal_ids_[2] = 1;
temporal_ids_[3] = 2;
cfg->ts_number_layers = number_of_temporal_layers_;
cfg->ts_periodicity = temporal_ids_length_;
// Split stream 40% 20% 40%.
// Bitrate API for VP8 is the agregated bitrate for all lower layers.
cfg->ts_target_bitrate[0] = bitrateKbit * kVp8LayerRateAlloction[2][0];
cfg->ts_target_bitrate[1] = bitrateKbit * kVp8LayerRateAlloction[2][1];
cfg->ts_target_bitrate[2] = bitrateKbit;
cfg->ts_rate_decimator[0] = 4;
cfg->ts_rate_decimator[1] = 2;
cfg->ts_rate_decimator[2] = 1;
memcpy(cfg->ts_layer_id,
temporal_ids_,
sizeof(unsigned int) * temporal_ids_length_);
temporal_pattern_length_ = 8;
temporal_pattern_[0] = kTemporalUpdateLastAndGoldenRefAltRef;
temporal_pattern_[1] = kTemporalUpdateNoneNoRefGoldenRefAltRef;
temporal_pattern_[2] = kTemporalUpdateGoldenWithoutDependencyRefAltRef;
temporal_pattern_[3] = kTemporalUpdateNone;
temporal_pattern_[4] = kTemporalUpdateLastRefAltRef;
temporal_pattern_[5] = kTemporalUpdateNone;
temporal_pattern_[6] = kTemporalUpdateGoldenRefAltRef;
temporal_pattern_[7] = kTemporalUpdateNone;
break;
case 4:
temporal_ids_length_ = 8;
temporal_ids_[0] = 0;
temporal_ids_[1] = 3;
temporal_ids_[2] = 2;
temporal_ids_[3] = 3;
temporal_ids_[4] = 1;
temporal_ids_[5] = 3;
temporal_ids_[6] = 2;
temporal_ids_[7] = 3;
// Split stream 25% 15% 20% 40%.
// Bitrate API for VP8 is the agregated bitrate for all lower layers.
cfg->ts_number_layers = 4;
cfg->ts_periodicity = temporal_ids_length_;
cfg->ts_target_bitrate[0] = bitrateKbit * kVp8LayerRateAlloction[3][0];
cfg->ts_target_bitrate[1] = bitrateKbit * kVp8LayerRateAlloction[3][1];
cfg->ts_target_bitrate[2] = bitrateKbit * kVp8LayerRateAlloction[3][2];
cfg->ts_target_bitrate[3] = bitrateKbit;
cfg->ts_rate_decimator[0] = 8;
cfg->ts_rate_decimator[1] = 4;
cfg->ts_rate_decimator[2] = 2;
cfg->ts_rate_decimator[3] = 1;
memcpy(cfg->ts_layer_id,
temporal_ids_,
sizeof(unsigned int) * temporal_ids_length_);
temporal_pattern_length_ = 16;
temporal_pattern_[0] = kTemporalUpdateLast;
temporal_pattern_[1] = kTemporalUpdateNone;
temporal_pattern_[2] = kTemporalUpdateAltrefWithoutDependency;
temporal_pattern_[3] = kTemporalUpdateNone;
temporal_pattern_[4] = kTemporalUpdateGoldenWithoutDependency;
temporal_pattern_[5] = kTemporalUpdateNone;
temporal_pattern_[6] = kTemporalUpdateAltref;
temporal_pattern_[7] = kTemporalUpdateNone;
temporal_pattern_[8] = kTemporalUpdateLast;
temporal_pattern_[9] = kTemporalUpdateNone;
temporal_pattern_[10] = kTemporalUpdateAltref;
temporal_pattern_[11] = kTemporalUpdateNone;
temporal_pattern_[12] = kTemporalUpdateGolden;
temporal_pattern_[13] = kTemporalUpdateNone;
temporal_pattern_[14] = kTemporalUpdateAltref;
temporal_pattern_[15] = kTemporalUpdateNone;
break;
default:
assert(false);
return false;
}
return true;
}
int TemporalLayers::EncodeFlags() {
assert(number_of_temporal_layers_ > 1);
assert(kMaxTemporalPattern >= temporal_pattern_length_);
assert(0 < temporal_pattern_length_);
int flags = 0;
int patternIdx = ++pattern_idx_ % temporal_pattern_length_;
assert(kMaxTemporalPattern >= patternIdx);
switch (temporal_pattern_[patternIdx]) {
case kTemporalUpdateLast:
flags |= VP8_EFLAG_NO_UPD_GF;
flags |= VP8_EFLAG_NO_UPD_ARF;
flags |= VP8_EFLAG_NO_REF_GF;
flags |= VP8_EFLAG_NO_REF_ARF;
break;
case kTemporalUpdateGoldenWithoutDependency:
flags |= VP8_EFLAG_NO_REF_GF;
// Deliberately no break here.
case kTemporalUpdateGolden:
flags |= VP8_EFLAG_NO_REF_ARF;
flags |= VP8_EFLAG_NO_UPD_ARF;
flags |= VP8_EFLAG_NO_UPD_LAST;
break;
case kTemporalUpdateAltrefWithoutDependency:
flags |= VP8_EFLAG_NO_REF_ARF;
flags |= VP8_EFLAG_NO_REF_GF;
// Deliberately no break here.
case kTemporalUpdateAltref:
flags |= VP8_EFLAG_NO_UPD_GF;
flags |= VP8_EFLAG_NO_UPD_LAST;
break;
case kTemporalUpdateNoneNoRefAltref:
flags |= VP8_EFLAG_NO_REF_ARF;
// Deliberately no break here.
case kTemporalUpdateNone:
flags |= VP8_EFLAG_NO_UPD_GF;
flags |= VP8_EFLAG_NO_UPD_ARF;
flags |= VP8_EFLAG_NO_UPD_LAST;
flags |= VP8_EFLAG_NO_UPD_ENTROPY;
break;
case kTemporalUpdateNoneNoRefGoldenRefAltRef:
flags |= VP8_EFLAG_NO_REF_GF;
flags |= VP8_EFLAG_NO_UPD_GF;
flags |= VP8_EFLAG_NO_UPD_ARF;
flags |= VP8_EFLAG_NO_UPD_LAST;
flags |= VP8_EFLAG_NO_UPD_ENTROPY;
break;
case kTemporalUpdateGoldenWithoutDependencyRefAltRef:
flags |= VP8_EFLAG_NO_REF_GF;
flags |= VP8_EFLAG_NO_UPD_ARF;
flags |= VP8_EFLAG_NO_UPD_LAST;
break;
case kTemporalUpdateLastRefAltRef:
flags |= VP8_EFLAG_NO_UPD_GF;
flags |= VP8_EFLAG_NO_UPD_ARF;
flags |= VP8_EFLAG_NO_REF_GF;
break;
case kTemporalUpdateGoldenRefAltRef:
flags |= VP8_EFLAG_NO_UPD_ARF;
flags |= VP8_EFLAG_NO_UPD_LAST;
break;
case kTemporalUpdateLastAndGoldenRefAltRef:
flags |= VP8_EFLAG_NO_UPD_ARF;
flags |= VP8_EFLAG_NO_REF_GF;
break;
}
return flags;
}
void TemporalLayers::PopulateCodecSpecific(bool key_frame,
CodecSpecificInfoVP8 *vp8_info,
uint32_t timestamp) {
assert(number_of_temporal_layers_ > 1);
assert(0 < temporal_ids_length_);
if (key_frame) {
// Keyframe is always temporal layer 0
vp8_info->temporalIdx = 0;
} else {
vp8_info->temporalIdx = temporal_ids_[pattern_idx_ % temporal_ids_length_];
}
TemporalReferences temporal_reference =
temporal_pattern_[pattern_idx_ % temporal_pattern_length_];
if (temporal_reference == kTemporalUpdateAltrefWithoutDependency ||
temporal_reference == kTemporalUpdateGoldenWithoutDependency ||
temporal_reference == kTemporalUpdateGoldenWithoutDependencyRefAltRef ||
temporal_reference == kTemporalUpdateNoneNoRefGoldenRefAltRef ||
(temporal_reference == kTemporalUpdateNone &&
number_of_temporal_layers_ == 4)) {
vp8_info->layerSync = true;
} else {
vp8_info->layerSync = false;
}
if (vp8_info->temporalIdx == 0 && timestamp != timestamp_) {
timestamp_ = timestamp;
tl0_pic_idx_++;
}
vp8_info->tl0PicIdx = tl0_pic_idx_;
}
} // namespace webrtc

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/* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* This file defines classes for doing temporal layers with VP8.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_TEMPORAL_LAYERS_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_TEMPORAL_LAYERS_H_
#include <typedefs.h>
// VPX forward declaration
typedef struct vpx_codec_enc_cfg vpx_codec_enc_cfg_t;
namespace webrtc {
struct CodecSpecificInfoVP8;
class TemporalLayers {
public:
TemporalLayers(int number_of_temporal_layers);
// Returns the recommended VP8 encode flags needed. May refresh the decoder
// and/or update the reference buffers.
int EncodeFlags();
bool ConfigureBitrates(int bitrate_kbit, vpx_codec_enc_cfg_t* cfg);
void PopulateCodecSpecific(bool key_frame, CodecSpecificInfoVP8 *vp8_info,
uint32_t timestamp);
private:
enum TemporalReferences {
// First base layer frame for 3 temporal layers, which updates last and
// golden with alt ref dependency.
kTemporalUpdateLastAndGoldenRefAltRef = 11,
// First enhancement layer with alt ref dependency.
kTemporalUpdateGoldenRefAltRef = 10,
// First enhancement layer with alt ref dependency.
kTemporalUpdateGoldenWithoutDependencyRefAltRef = 9,
// Base layer with alt ref dependency.
kTemporalUpdateLastRefAltRef = 8,
// Highest enhacement layer without dependency on golden with alt ref
// dependency.
kTemporalUpdateNoneNoRefGoldenRefAltRef = 7,
// Second layer and last frame in cycle, for 2 layers.
kTemporalUpdateNoneNoRefAltref = 6,
// Highest enhancement layer.
kTemporalUpdateNone = 5,
// Second enhancement layer.
kTemporalUpdateAltref = 4,
// Second enhancement layer without dependency on previous frames in
// the second enhancement layer.
kTemporalUpdateAltrefWithoutDependency = 3,
// First enhancement layer.
kTemporalUpdateGolden = 2,
// First enhancement layer without dependency on previous frames in
// the first enhancement layer.
kTemporalUpdateGoldenWithoutDependency = 1,
// Base layer.
kTemporalUpdateLast = 0,
};
enum { kMaxTemporalPattern = 16 };
int number_of_temporal_layers_;
int temporal_ids_length_;
int temporal_ids_[kMaxTemporalPattern];
int temporal_pattern_length_;
TemporalReferences temporal_pattern_[kMaxTemporalPattern];
uint8_t tl0_pic_idx_;
uint8_t pattern_idx_;
uint32_t timestamp_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_TEMPORAL_LAYERS_H_

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "gtest/gtest.h"
#include "temporal_layers.h"
#include "video_codec_interface.h"
#include "vpx/vpx_encoder.h"
#include "vpx/vp8cx.h"
namespace webrtc {
enum {
kTemporalUpdateLast = VP8_EFLAG_NO_UPD_GF |
VP8_EFLAG_NO_UPD_ARF |
VP8_EFLAG_NO_REF_GF |
VP8_EFLAG_NO_REF_ARF,
kTemporalUpdateGoldenWithoutDependency = VP8_EFLAG_NO_REF_GF |
VP8_EFLAG_NO_REF_ARF |
VP8_EFLAG_NO_UPD_ARF |
VP8_EFLAG_NO_UPD_LAST,
kTemporalUpdateGolden = VP8_EFLAG_NO_REF_ARF |
VP8_EFLAG_NO_UPD_ARF |
VP8_EFLAG_NO_UPD_LAST,
kTemporalUpdateAltrefWithoutDependency = VP8_EFLAG_NO_REF_ARF |
VP8_EFLAG_NO_REF_GF |
VP8_EFLAG_NO_UPD_GF |
VP8_EFLAG_NO_UPD_LAST,
kTemporalUpdateAltref = VP8_EFLAG_NO_UPD_GF |
VP8_EFLAG_NO_UPD_LAST,
kTemporalUpdateNone = VP8_EFLAG_NO_UPD_GF |
VP8_EFLAG_NO_UPD_ARF |
VP8_EFLAG_NO_UPD_LAST |
VP8_EFLAG_NO_UPD_ENTROPY,
kTemporalUpdateNoneNoRefAltRef = VP8_EFLAG_NO_REF_ARF |
VP8_EFLAG_NO_UPD_GF |
VP8_EFLAG_NO_UPD_ARF |
VP8_EFLAG_NO_UPD_LAST |
VP8_EFLAG_NO_UPD_ENTROPY,
kTemporalUpdateNoneNoRefGolden = VP8_EFLAG_NO_REF_GF |
VP8_EFLAG_NO_UPD_GF |
VP8_EFLAG_NO_UPD_ARF |
VP8_EFLAG_NO_UPD_LAST |
VP8_EFLAG_NO_UPD_ENTROPY,
kTemporalUpdateGoldenWithoutDependencyRefAltRef = VP8_EFLAG_NO_REF_GF |
VP8_EFLAG_NO_UPD_ARF |
VP8_EFLAG_NO_UPD_LAST,
kTemporalUpdateGoldenRefAltRef = VP8_EFLAG_NO_UPD_ARF |
VP8_EFLAG_NO_UPD_LAST,
kTemporalUpdateLastRefAltRef = VP8_EFLAG_NO_UPD_GF |
VP8_EFLAG_NO_UPD_ARF |
VP8_EFLAG_NO_REF_GF,
kTemporalUpdateLastAndGoldenRefAltRef = VP8_EFLAG_NO_UPD_ARF |
VP8_EFLAG_NO_REF_GF,
};
TEST(TemporalLayersTest, 2Layers) {
TemporalLayers tl(2);
vpx_codec_enc_cfg_t cfg;
CodecSpecificInfoVP8 vp8_info;
tl.ConfigureBitrates(500, &cfg);
int expected_flags[16] = { kTemporalUpdateLastAndGoldenRefAltRef,
kTemporalUpdateGoldenWithoutDependencyRefAltRef,
kTemporalUpdateLastRefAltRef,
kTemporalUpdateGoldenRefAltRef,
kTemporalUpdateLastRefAltRef,
kTemporalUpdateGoldenRefAltRef,
kTemporalUpdateLastRefAltRef,
kTemporalUpdateNone,
kTemporalUpdateLastAndGoldenRefAltRef,
kTemporalUpdateGoldenWithoutDependencyRefAltRef,
kTemporalUpdateLastRefAltRef,
kTemporalUpdateGoldenRefAltRef,
kTemporalUpdateLastRefAltRef,
kTemporalUpdateGoldenRefAltRef,
kTemporalUpdateLastRefAltRef,
kTemporalUpdateNone,
};
int expected_temporal_idx[16] =
{ 0, 1, 0, 1, 0, 1, 0, 1, 0, 1, 0, 1, 0, 1, 0, 1 };
bool expected_layer_sync[16] =
{ false, true, false, false, false, false, false, false,
false, true, false, false, false, false, false, false };
for (int i = 0; i < 16; ++i) {
EXPECT_EQ(expected_flags[i], tl.EncodeFlags());
tl.PopulateCodecSpecific(false, &vp8_info, 0);
EXPECT_EQ(expected_temporal_idx[i], vp8_info.temporalIdx);
EXPECT_EQ(expected_layer_sync[i], vp8_info.layerSync);
}
}
TEST(TemporalLayersTest, 3Layers) {
TemporalLayers tl(3);
vpx_codec_enc_cfg_t cfg;
CodecSpecificInfoVP8 vp8_info;
tl.ConfigureBitrates(500, &cfg);
int expected_flags[16] = { kTemporalUpdateLastAndGoldenRefAltRef,
kTemporalUpdateNoneNoRefGolden,
kTemporalUpdateGoldenWithoutDependencyRefAltRef,
kTemporalUpdateNone,
kTemporalUpdateLastRefAltRef,
kTemporalUpdateNone,
kTemporalUpdateGoldenRefAltRef,
kTemporalUpdateNone,
kTemporalUpdateLastAndGoldenRefAltRef,
kTemporalUpdateNoneNoRefGolden,
kTemporalUpdateGoldenWithoutDependencyRefAltRef,
kTemporalUpdateNone,
kTemporalUpdateLastRefAltRef,
kTemporalUpdateNone,
kTemporalUpdateGoldenRefAltRef,
kTemporalUpdateNone,
};
int expected_temporal_idx[16] =
{ 0, 2, 1, 2, 0, 2, 1, 2, 0, 2, 1, 2, 0, 2, 1, 2 };
bool expected_layer_sync[16] =
{ false, true, true, false, false, false, false, false,
false, true, true, false, false, false, false, false };
for (int i = 0; i < 16; ++i) {
EXPECT_EQ(expected_flags[i], tl.EncodeFlags());
tl.PopulateCodecSpecific(false, &vp8_info, 0);
EXPECT_EQ(expected_temporal_idx[i], vp8_info.temporalIdx);
EXPECT_EQ(expected_layer_sync[i], vp8_info.layerSync);
}
}
TEST(TemporalLayersTest, 4Layers) {
TemporalLayers tl(4);
vpx_codec_enc_cfg_t cfg;
CodecSpecificInfoVP8 vp8_info;
tl.ConfigureBitrates(500, &cfg);
int expected_flags[16] = {
kTemporalUpdateLast,
kTemporalUpdateNone,
kTemporalUpdateAltrefWithoutDependency,
kTemporalUpdateNone,
kTemporalUpdateGoldenWithoutDependency,
kTemporalUpdateNone,
kTemporalUpdateAltref,
kTemporalUpdateNone,
kTemporalUpdateLast,
kTemporalUpdateNone,
kTemporalUpdateAltref,
kTemporalUpdateNone,
kTemporalUpdateGolden,
kTemporalUpdateNone,
kTemporalUpdateAltref,
kTemporalUpdateNone,
};
int expected_temporal_idx[16] =
{ 0, 3, 2, 3, 1, 3, 2, 3, 0, 3, 2, 3, 1, 3, 2, 3 };
bool expected_layer_sync[16] =
{ false, true, true, true, true, true, false, true,
false, true, false, true, false, true, false, true };
for (int i = 0; i < 16; ++i) {
EXPECT_EQ(expected_flags[i], tl.EncodeFlags());
tl.PopulateCodecSpecific(false, &vp8_info, 0);
EXPECT_EQ(expected_temporal_idx[i], vp8_info.temporalIdx);
EXPECT_EQ(expected_layer_sync[i], vp8_info.layerSync);
}
}
TEST(TemporalLayersTest, KeyFrame) {
TemporalLayers tl(3);
vpx_codec_enc_cfg_t cfg;
CodecSpecificInfoVP8 vp8_info;
tl.ConfigureBitrates(500, &cfg);
int expected_flags[8] = {
kTemporalUpdateLastAndGoldenRefAltRef,
kTemporalUpdateNoneNoRefGolden,
kTemporalUpdateGoldenWithoutDependencyRefAltRef,
kTemporalUpdateNone,
kTemporalUpdateLastRefAltRef,
kTemporalUpdateNone,
kTemporalUpdateGoldenRefAltRef,
kTemporalUpdateNone,
};
int expected_temporal_idx[8] =
{ 0, 0, 0, 0, 0, 0, 0, 2};
bool expected_layer_sync[8] =
{ false, true, true, false, false, false, false, false };
for (int i = 0; i < 7; ++i) {
EXPECT_EQ(expected_flags[i], tl.EncodeFlags());
tl.PopulateCodecSpecific(true, &vp8_info, 0);
EXPECT_EQ(expected_temporal_idx[i], vp8_info.temporalIdx);
EXPECT_EQ(expected_layer_sync[i], vp8_info.layerSync);
}
EXPECT_EQ(expected_flags[7], tl.EncodeFlags());
tl.PopulateCodecSpecific(false, &vp8_info, 0);
EXPECT_EQ(expected_temporal_idx[7], vp8_info.temporalIdx);
EXPECT_EQ(expected_layer_sync[7], vp8_info.layerSync);
}
} // namespace webrtc

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "benchmark.h"
#include "testsupport/fileutils.h"
#include "vp8.h"
using namespace webrtc;
VP8Benchmark::VP8Benchmark()
: Benchmark("VP8Benchmark", "VP8 benchmark over a range of test cases",
webrtc::test::OutputPath() + "VP8Benchmark.txt", "VP8") {
}
VP8Benchmark::VP8Benchmark(std::string name, std::string description)
: Benchmark(name, description,
webrtc::test::OutputPath() + "VP8Benchmark.txt",
"VP8") {
}
VP8Benchmark::VP8Benchmark(std::string name, std::string description,
std::string resultsFileName)
: Benchmark(name, description, resultsFileName, "VP8") {
}
VideoEncoder* VP8Benchmark::GetNewEncoder() {
return VP8Encoder::Create();
}
VideoDecoder* VP8Benchmark::GetNewDecoder() {
return VP8Decoder::Create();
}

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_BENCHMARK_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_BENCHMARK_H_
#include "modules/video_coding/codecs/test_framework/benchmark.h"
class VP8Benchmark : public Benchmark
{
public:
VP8Benchmark();
VP8Benchmark(std::string name, std::string description);
VP8Benchmark(std::string name, std::string description, std::string resultsFileName);
protected:
virtual webrtc::VideoEncoder* GetNewEncoder();
virtual webrtc::VideoDecoder* GetNewDecoder();
};
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_BENCHMARK_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "dual_decoder_test.h"
#include <assert.h>
#include <string.h> // memcmp
#include <time.h>
#include "testsupport/fileutils.h"
VP8DualDecoderTest::VP8DualDecoderTest(float bitRate)
:
VP8NormalAsyncTest(bitRate)
{
_decoder2 = NULL;
}
VP8DualDecoderTest::VP8DualDecoderTest()
:
VP8NormalAsyncTest("VP8 Dual Decoder Test", "Tests VP8 dual decoder", 1),
_decoder2(NULL)
{}
VP8DualDecoderTest::~VP8DualDecoderTest()
{
if(_decoder2)
{
_decoder2->Release();
delete _decoder2;
}
_decodedVideoBuffer2.Free();
}
void
VP8DualDecoderTest::Perform()
{
_inname = webrtc::test::ProjectRootPath() + "resources/foreman_cif.yuv";
CodecSettings(352, 288, 30, _bitRate);
Setup();
_inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
_decodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
_decodedVideoBuffer2.VerifyAndAllocate(_lengthSourceFrame);
if(_encoder->InitEncode(&_inst, 4, 1460) < 0)
{
exit(EXIT_FAILURE);
}
_decoder->InitDecode(&_inst,1);
FrameQueue frameQueue;
VideoEncodeCompleteCallback encCallback(_encodedFile, &frameQueue, *this);
DualDecoderCompleteCallback decCallback(&_decodedVideoBuffer);
DualDecoderCompleteCallback decCallback2(&_decodedVideoBuffer2);
_encoder->RegisterEncodeCompleteCallback(&encCallback);
_decoder->RegisterDecodeCompleteCallback(&decCallback);
if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
{
exit(EXIT_FAILURE);
}
_totalEncodeTime = _totalDecodeTime = 0;
_totalEncodePipeTime = _totalDecodePipeTime = 0;
bool complete = false;
_framecnt = 0;
_encFrameCnt = 0;
_decFrameCnt = 0;
_sumEncBytes = 0;
_lengthEncFrame = 0;
double starttime = clock()/(double)CLOCKS_PER_SEC;
while (!complete)
{
if (_encFrameCnt == 10)
{
// initialize second decoder and copy state
_decoder2 = static_cast<webrtc::VP8Decoder *>(_decoder->Copy());
assert(_decoder2 != NULL);
_decoder2->RegisterDecodeCompleteCallback(&decCallback2);
}
CodecSpecific_InitBitrate();
complete = Encode();
if (!frameQueue.Empty() || complete)
{
while (!frameQueue.Empty())
{
_frameToDecode =
static_cast<FrameQueueTuple *>(frameQueue.PopFrame());
int lost = DoPacketLoss();
if (lost == 2)
{
// Lost the whole frame, continue
_missingFrames = true;
delete _frameToDecode;
_frameToDecode = NULL;
continue;
}
int ret = Decode(lost);
delete _frameToDecode;
_frameToDecode = NULL;
if (ret < 0)
{
fprintf(stderr,"\n\nError in decoder: %d\n\n", ret);
exit(EXIT_FAILURE);
}
else if (ret == 0)
{
_framecnt++;
}
else
{
fprintf(stderr,
"\n\nPositive return value from decode!\n\n");
}
}
}
}
double endtime = clock()/(double)CLOCKS_PER_SEC;
double totalExecutionTime = endtime - starttime;
printf("Total execution time: %.1f s\n", totalExecutionTime);
_sumEncBytes = encCallback.EncodedBytes();
double actualBitRate = ActualBitRate(_encFrameCnt) / 1000.0;
double avgEncTime = _totalEncodeTime / _encFrameCnt;
double avgDecTime = _totalDecodeTime / _decFrameCnt;
printf("Actual bitrate: %f kbps\n", actualBitRate);
printf("Average encode time: %.1f ms\n", 1000 * avgEncTime);
printf("Average decode time: %.1f ms\n", 1000 * avgDecTime);
printf("Average encode pipeline time: %.1f ms\n",
1000 * _totalEncodePipeTime / _encFrameCnt);
printf("Average decode pipeline time: %.1f ms\n",
1000 * _totalDecodePipeTime / _decFrameCnt);
printf("Number of encoded frames: %u\n", _encFrameCnt);
printf("Number of decoded frames: %u\n", _decFrameCnt);
(*_log) << "Actual bitrate: " << actualBitRate << " kbps\tTarget: " <<
_bitRate << " kbps" << std::endl;
(*_log) << "Average encode time: " << avgEncTime << " s" << std::endl;
(*_log) << "Average decode time: " << avgDecTime << " s" << std::endl;
_encoder->Release();
_decoder->Release();
Teardown();
}
int
VP8DualDecoderTest::Decode(int lossValue)
{
_sumEncBytes += _frameToDecode->_frame->Length();
webrtc::EncodedImage encodedImage;
VideoEncodedBufferToEncodedImage(*(_frameToDecode->_frame), encodedImage);
encodedImage._completeFrame = !lossValue;
_decodeCompleteTime = 0;
_decodeTimes[encodedImage._timeStamp] = clock()/(double)CLOCKS_PER_SEC;
int ret = _decoder->Decode(encodedImage, _missingFrames, NULL,
_frameToDecode->_codecSpecificInfo);
// second decoder
if (_decoder2)
{
int ret2 = _decoder2->Decode(encodedImage, _missingFrames, NULL,
_frameToDecode->_codecSpecificInfo,
0 /* dummy */);
// check return values
if (ret < 0 || ret2 < 0 || ret2 != ret)
{
exit(EXIT_FAILURE);
}
// compare decoded images
if (!CheckIfBitExact(_decodedVideoBuffer.Buffer(),
_decodedVideoBuffer.Length(),
_decodedVideoBuffer2.Buffer(), _decodedVideoBuffer.Length()))
{
fprintf(stderr,"\n\nClone output different from master.\n\n");
exit(EXIT_FAILURE);
}
}
_missingFrames = false;
return ret;
}
bool
VP8DualDecoderTest::CheckIfBitExact(const void* ptrA, unsigned int aLengthBytes,
const void* ptrB, unsigned int bLengthBytes)
{
if (aLengthBytes != bLengthBytes)
{
return false;
}
return memcmp(ptrA, ptrB, aLengthBytes) == 0;
}
WebRtc_Word32 DualDecoderCompleteCallback::Decoded(webrtc::VideoFrame& image)
{
_decodedVideoBuffer->VerifyAndAllocate(image.Length());
_decodedVideoBuffer->CopyFrame(image.Length(), image.Buffer());
_decodedVideoBuffer->SetWidth(image.Width());
_decodedVideoBuffer->SetHeight(image.Height());
_decodedVideoBuffer->SetTimeStamp(image.TimeStamp());
_decodeComplete = true;
return 0;
}
bool DualDecoderCompleteCallback::DecodeComplete()
{
if (_decodeComplete)
{
_decodeComplete = false;
return true;
}
return false;
}

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_DUAL_DECODER_TEST_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_DUAL_DECODER_TEST_H_
#include "vp8.h"
#include "normal_async_test.h"
class DualDecoderCompleteCallback;
class VP8DualDecoderTest : public VP8NormalAsyncTest
{
public:
VP8DualDecoderTest(float bitRate);
VP8DualDecoderTest();
virtual ~VP8DualDecoderTest();
virtual void Perform();
protected:
VP8DualDecoderTest(std::string name, std::string description,
unsigned int testNo)
: VP8NormalAsyncTest(name, description, testNo) {}
virtual int Decode(int lossValue = 0);
webrtc::VP8Decoder* _decoder2;
webrtc::VideoFrame _decodedVideoBuffer2;
static bool CheckIfBitExact(const void *ptrA, unsigned int aLengthBytes,
const void *ptrB, unsigned int bLengthBytes);
private:
};
class DualDecoderCompleteCallback : public webrtc::DecodedImageCallback
{
public:
DualDecoderCompleteCallback(webrtc::VideoFrame* buffer)
: _decodedVideoBuffer(buffer), _decodeComplete(false) {}
WebRtc_Word32 Decoded(webrtc::VideoFrame& decodedImage);
bool DecodeComplete();
private:
webrtc::VideoFrame* _decodedVideoBuffer;
bool _decodeComplete;
};
#endif

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "normal_async_test.h"
using namespace webrtc;
VP8NormalAsyncTest::VP8NormalAsyncTest(WebRtc_UWord32 bitRate) :
NormalAsyncTest("VP8 Normal Test 1", "Tests VP8 normal execution", bitRate, 1),
_hasReceivedRPSI(false)
{
}
VP8NormalAsyncTest::VP8NormalAsyncTest(WebRtc_UWord32 bitRate, unsigned int testNo):
NormalAsyncTest("VP8 Normal Test 1", "Tests VP8 normal execution", bitRate, testNo),
_hasReceivedRPSI(false)
{
}
void
VP8NormalAsyncTest::CodecSettings(int width, int height, WebRtc_UWord32 frameRate /*=30*/, WebRtc_UWord32 bitRate /*=0*/)
{
if (bitRate > 0)
{
_bitRate = bitRate;
}else if (_bitRate == 0)
{
_bitRate = 600;
}
_inst.codecType = kVideoCodecVP8;
_inst.codecSpecific.VP8.feedbackModeOn = true;
_inst.codecSpecific.VP8.pictureLossIndicationOn = true;
_inst.codecSpecific.VP8.complexity = kComplexityNormal;
_inst.maxFramerate = (unsigned char)frameRate;
_inst.startBitrate = _bitRate;
_inst.maxBitrate = 8000;
_inst.width = width;
_inst.height = height;
}
void
VP8NormalAsyncTest::CodecSpecific_InitBitrate()
{
if (_bitRate == 0)
{
_encoder->SetRates(600, _inst.maxFramerate);
}else
{
_encoder->SetRates(_bitRate, _inst.maxFramerate);
}
}
WebRtc_Word32
VP8NormalAsyncTest::ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId)
{
_pictureIdRPSI = pictureId;
_hasReceivedRPSI = true;
return 0;
}
CodecSpecificInfo*
VP8NormalAsyncTest::CreateEncoderSpecificInfo() const
{
CodecSpecificInfo* vp8CodecSpecificInfo = new CodecSpecificInfo();
vp8CodecSpecificInfo->codecType = kVideoCodecVP8;
vp8CodecSpecificInfo->codecSpecific.VP8.hasReceivedRPSI = _hasReceivedRPSI;
vp8CodecSpecificInfo->codecSpecific.VP8.pictureIdRPSI = _pictureIdRPSI;
vp8CodecSpecificInfo->codecSpecific.VP8.hasReceivedSLI = _hasReceivedSLI;
vp8CodecSpecificInfo->codecSpecific.VP8.pictureIdSLI = _pictureIdSLI;
_hasReceivedSLI = false;
_hasReceivedRPSI = false;
return vp8CodecSpecificInfo;
}

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_NORMAL_ASYNC_TEST_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_NORMAL_ASYNC_TEST_H_
#include "modules/video_coding/codecs/test_framework/normal_async_test.h"
class VP8NormalAsyncTest : public NormalAsyncTest
{
public:
VP8NormalAsyncTest(WebRtc_UWord32 bitRate);
VP8NormalAsyncTest(WebRtc_UWord32 bitRate, unsigned int testNo);
VP8NormalAsyncTest() : NormalAsyncTest("VP8 Normal Test 1", "Tests VP8 normal execution", 1) {}
protected:
VP8NormalAsyncTest(std::string name, std::string description, unsigned int testNo) : NormalAsyncTest(name, description, testNo) {}
virtual void CodecSpecific_InitBitrate();
virtual void CodecSettings(int width, int height, WebRtc_UWord32 frameRate=30, WebRtc_UWord32 bitRate=0);
virtual webrtc::CodecSpecificInfo* CreateEncoderSpecificInfo() const;
virtual WebRtc_Word32 ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId);
private:
mutable bool _hasReceivedRPSI;
WebRtc_UWord64 _pictureIdRPSI;
};
#endif

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "packet_loss_test.h"
#include <cassert>
VP8PacketLossTest::VP8PacketLossTest()
:
PacketLossTest("VP8PacketLossTest", "Encode, remove lost packets, decode")
{
}
VP8PacketLossTest::VP8PacketLossTest(std::string name, std::string description)
:
PacketLossTest(name, description)
{
}
VP8PacketLossTest::VP8PacketLossTest(double lossRate,
bool useNack,
int rttFrames)
:
PacketLossTest("VP8PacketLossTest", "Encode, remove lost packets, decode",
lossRate, useNack, rttFrames)
{
}
int VP8PacketLossTest::ByteLoss(int size, unsigned char* /* pkg */, int bytesToLose)
{
int retLength = size - bytesToLose;
if (retLength < 4)
{
retLength = 4;
}
return retLength;
}
WebRtc_Word32
VP8PacketLossTest::ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId)
{
_pictureIdRPSI = pictureId;
_hasReceivedRPSI = true;
return 0;
}
webrtc::CodecSpecificInfo*
VP8PacketLossTest::CreateEncoderSpecificInfo() const
{
webrtc::CodecSpecificInfo* vp8CodecSpecificInfo =
new webrtc::CodecSpecificInfo();
vp8CodecSpecificInfo->codecType = webrtc::kVideoCodecVP8;
vp8CodecSpecificInfo->codecSpecific.VP8.hasReceivedRPSI = _hasReceivedRPSI;
vp8CodecSpecificInfo->codecSpecific.VP8.pictureIdRPSI = _pictureIdRPSI;
vp8CodecSpecificInfo->codecSpecific.VP8.hasReceivedSLI = _hasReceivedSLI;
vp8CodecSpecificInfo->codecSpecific.VP8.pictureIdSLI = _pictureIdSLI;
_hasReceivedSLI = false;
_hasReceivedRPSI = false;
return vp8CodecSpecificInfo;
}
bool VP8PacketLossTest::PacketLoss(double lossRate, int numLosses) {
if (numLosses)
return true;
return RandUniform() < lossRate;
}

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_PACKET_LOSS_TEST_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_PACKET_LOSS_TEST_H_
#include "modules/video_coding/codecs/test_framework/packet_loss_test.h"
class VP8PacketLossTest : public PacketLossTest
{
public:
VP8PacketLossTest();
VP8PacketLossTest(double lossRate, bool useNack, int rttFrames);
protected:
VP8PacketLossTest(std::string name, std::string description);
virtual int ByteLoss(int size, unsigned char *pkg, int bytesToLose);
WebRtc_Word32 ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId);
// |lossRate| is the probability of packet loss between 0 and 1.
// |numLosses| is the number of packets already lost in the current frame.
virtual bool PacketLoss(double lossRate, int numLosses);
webrtc::CodecSpecificInfo* CreateEncoderSpecificInfo() const;
};
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_PACKET_LOSS_TEST_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "rps_test.h"
#include <assert.h>
#include <string.h> // memcmp
#include <time.h>
#include "vp8.h"
VP8RpsTest::VP8RpsTest(float bitRate)
: VP8NormalAsyncTest(bitRate),
decoder2_(webrtc::VP8Decoder::Create()),
sli_(false) {
}
VP8RpsTest::VP8RpsTest()
: VP8NormalAsyncTest("VP8 Reference Picture Selection Test",
"VP8 Reference Picture Selection Test", 1),
decoder2_(webrtc::VP8Decoder::Create()),
sli_(false) {
}
VP8RpsTest::~VP8RpsTest() {
if (decoder2_) {
decoder2_->Release();
delete decoder2_;
}
decoded_frame2_.Free();
}
void VP8RpsTest::Perform() {
_inname = "test/testFiles/foreman_cif.yuv";
CodecSettings(352, 288, 30, _bitRate);
Setup();
_inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
_decodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
decoded_frame2_.VerifyAndAllocate(_lengthSourceFrame);
// Enable RPS functionality
_inst.codecSpecific.VP8.pictureLossIndicationOn = true;
_inst.codecSpecific.VP8.feedbackModeOn = true;
if(_encoder->InitEncode(&_inst, 4, 1460) < 0)
exit(EXIT_FAILURE);
_decoder->InitDecode(&_inst,1);
decoder2_->InitDecode(&_inst,1);
FrameQueue frameQueue;
VideoEncodeCompleteCallback encCallback(_encodedFile, &frameQueue, *this);
RpsDecodeCompleteCallback decCallback(&_decodedVideoBuffer);
RpsDecodeCompleteCallback decCallback2(&decoded_frame2_);
_encoder->RegisterEncodeCompleteCallback(&encCallback);
_decoder->RegisterDecodeCompleteCallback(&decCallback);
decoder2_->RegisterDecodeCompleteCallback(&decCallback2);
if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
exit(EXIT_FAILURE);
_totalEncodeTime = _totalDecodeTime = 0;
_totalEncodePipeTime = _totalDecodePipeTime = 0;
bool complete = false;
_framecnt = 0;
_encFrameCnt = 0;
_decFrameCnt = 0;
_sumEncBytes = 0;
_lengthEncFrame = 0;
double starttime = clock()/(double)CLOCKS_PER_SEC;
while (!complete) {
CodecSpecific_InitBitrate();
complete = EncodeRps(&decCallback2);
if (!frameQueue.Empty() || complete) {
while (!frameQueue.Empty()) {
_frameToDecode =
static_cast<FrameQueueTuple *>(frameQueue.PopFrame());
int lost = DoPacketLoss();
if (lost == 2) {
// Lost the whole frame, continue
_missingFrames = true;
delete _frameToDecode;
_frameToDecode = NULL;
continue;
}
int ret = Decode(lost);
delete _frameToDecode;
_frameToDecode = NULL;
if (ret < 0) {
fprintf(stderr,"\n\nError in decoder: %d\n\n", ret);
exit(EXIT_FAILURE);
}
else if (ret == 0) {
_framecnt++;
}
else {
fprintf(stderr,
"\n\nPositive return value from decode!\n\n");
}
}
}
}
double endtime = clock()/(double)CLOCKS_PER_SEC;
double totalExecutionTime = endtime - starttime;
printf("Total execution time: %.1f s\n", totalExecutionTime);
_sumEncBytes = encCallback.EncodedBytes();
double actualBitRate = ActualBitRate(_encFrameCnt) / 1000.0;
double avgEncTime = _totalEncodeTime / _encFrameCnt;
double avgDecTime = _totalDecodeTime / _decFrameCnt;
printf("Actual bitrate: %f kbps\n", actualBitRate);
printf("Average encode time: %.1f ms\n", 1000 * avgEncTime);
printf("Average decode time: %.1f ms\n", 1000 * avgDecTime);
printf("Average encode pipeline time: %.1f ms\n",
1000 * _totalEncodePipeTime / _encFrameCnt);
printf("Average decode pipeline time: %.1f ms\n",
1000 * _totalDecodePipeTime / _decFrameCnt);
printf("Number of encoded frames: %u\n", _encFrameCnt);
printf("Number of decoded frames: %u\n", _decFrameCnt);
(*_log) << "Actual bitrate: " << actualBitRate << " kbps\tTarget: " <<
_bitRate << " kbps" << std::endl;
(*_log) << "Average encode time: " << avgEncTime << " s" << std::endl;
(*_log) << "Average decode time: " << avgDecTime << " s" << std::endl;
_encoder->Release();
_decoder->Release();
Teardown();
}
bool VP8RpsTest::EncodeRps(RpsDecodeCompleteCallback* decodeCallback) {
_lengthEncFrame = 0;
size_t bytes_read = fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile);
if (bytes_read < _lengthSourceFrame)
return true;
_inputVideoBuffer.CopyFrame(_lengthSourceFrame, _sourceBuffer);
_inputVideoBuffer.SetTimeStamp((unsigned int)
(_encFrameCnt * 9e4 / _inst.maxFramerate));
_inputVideoBuffer.SetWidth(_inst.width);
_inputVideoBuffer.SetHeight(_inst.height);
if (feof(_sourceFile) != 0) {
return true;
}
_encodeCompleteTime = 0;
_encodeTimes[_inputVideoBuffer.TimeStamp()] = tGetTime();
webrtc::CodecSpecificInfo* codecSpecificInfo = CreateEncoderSpecificInfo();
codecSpecificInfo->codecSpecific.VP8.pictureIdRPSI =
decodeCallback->LastDecodedRefPictureId(
&codecSpecificInfo->codecSpecific.VP8.hasReceivedRPSI);
if (sli_) {
codecSpecificInfo->codecSpecific.VP8.pictureIdSLI =
decodeCallback->LastDecodedPictureId();
codecSpecificInfo->codecSpecific.VP8.hasReceivedSLI = true;
sli_ = false;
}
printf("Encoding: %u\n", _framecnt);
int ret = _encoder->Encode(_inputVideoBuffer, codecSpecificInfo, NULL);
if (ret < 0)
printf("Failed to encode: %u\n", _framecnt);
if (codecSpecificInfo != NULL) {
delete codecSpecificInfo;
codecSpecificInfo = NULL;
}
if (_encodeCompleteTime > 0) {
_totalEncodeTime += _encodeCompleteTime -
_encodeTimes[_inputVideoBuffer.TimeStamp()];
}
else {
_totalEncodeTime += tGetTime() -
_encodeTimes[_inputVideoBuffer.TimeStamp()];
}
return false;
}
//#define FRAME_LOSS 1
int VP8RpsTest::Decode(int lossValue) {
_sumEncBytes += _frameToDecode->_frame->Length();
webrtc::EncodedImage encodedImage;
VideoEncodedBufferToEncodedImage(*(_frameToDecode->_frame), encodedImage);
encodedImage._completeFrame = !lossValue;
_decodeCompleteTime = 0;
_decodeTimes[encodedImage._timeStamp] = clock()/(double)CLOCKS_PER_SEC;
int ret = _decoder->Decode(encodedImage, _missingFrames, NULL,
_frameToDecode->_codecSpecificInfo);
// Drop every 10th frame for the second decoder
#if FRAME_LOSS
if (_framecnt == 0 || _framecnt % 10 != 0) {
printf("Decoding: %u\n", _framecnt);
if (_framecnt > 1 && (_framecnt - 1) % 10 == 0)
_missingFrames = true;
#else
if (true) {
if (_framecnt > 0 && _framecnt % 10 == 0) {
encodedImage._length = std::rand() % encodedImage._length;
printf("Decoding with loss: %u\n", _framecnt);
}
else
printf("Decoding: %u\n", _framecnt);
#endif
int ret2 = decoder2_->Decode(encodedImage, _missingFrames, NULL,
_frameToDecode->_codecSpecificInfo,
0 /* dummy */);
// check return values
if (ret < 0 || ret2 < 0) {
return -1;
} else if (ret2 == WEBRTC_VIDEO_CODEC_ERR_REQUEST_SLI ||
ret2 == WEBRTC_VIDEO_CODEC_REQUEST_SLI) {
sli_ = true;
}
// compare decoded images
#if FRAME_LOSS
if (!_missingFrames) {
if (!CheckIfBitExact(_decodedVideoBuffer.GetBuffer(),
_decodedVideoBuffer.GetLength(),
decoded_frame2_.GetBuffer(), _decodedVideoBuffer.GetLength())) {
fprintf(stderr,"\n\nRPS decoder different from master: %u\n\n",
_framecnt);
return -1;
}
}
#else
if (_framecnt > 0 && _framecnt % 10 != 0) {
if (!CheckIfBitExact(_decodedVideoBuffer.Buffer(),
_decodedVideoBuffer.Length(),
decoded_frame2_.Buffer(), _decodedVideoBuffer.Length())) {
fprintf(stderr,"\n\nRPS decoder different from master: %u\n\n",
_framecnt);
return -1;
}
}
#endif
}
#if FRAME_LOSS
else
printf("Dropping %u\n", _framecnt);
#endif
_missingFrames = false;
return 0;
}
bool
VP8RpsTest::CheckIfBitExact(const void* ptrA, unsigned int aLengthBytes,
const void* ptrB, unsigned int bLengthBytes) {
if (aLengthBytes != bLengthBytes)
return false;
return memcmp(ptrA, ptrB, aLengthBytes) == 0;
}
RpsDecodeCompleteCallback::RpsDecodeCompleteCallback(webrtc::VideoFrame* buffer)
: decoded_frame_(buffer),
decode_complete_(false),
last_decoded_picture_id_(0),
last_decoded_ref_picture_id_(0),
updated_ref_picture_id_(false) {
}
WebRtc_Word32 RpsDecodeCompleteCallback::Decoded(webrtc::VideoFrame& image) {
return decoded_frame_->CopyFrame(image);
decode_complete_ = true;
}
bool RpsDecodeCompleteCallback::DecodeComplete() {
if (decode_complete_)
{
decode_complete_ = false;
return true;
}
return false;
}
WebRtc_Word32 RpsDecodeCompleteCallback::ReceivedDecodedReferenceFrame(
const WebRtc_UWord64 picture_id) {
last_decoded_ref_picture_id_ = picture_id & 0x7FFF;
updated_ref_picture_id_ = true;
return 0;
}
WebRtc_Word32 RpsDecodeCompleteCallback::ReceivedDecodedFrame(
const WebRtc_UWord64 picture_id) {
last_decoded_picture_id_ = picture_id & 0x3F;
return 0;
}
WebRtc_UWord64 RpsDecodeCompleteCallback::LastDecodedPictureId() const {
return last_decoded_picture_id_;
}
WebRtc_UWord64 RpsDecodeCompleteCallback::LastDecodedRefPictureId(
bool *updated) {
if (updated)
*updated = updated_ref_picture_id_;
updated_ref_picture_id_ = false;
return last_decoded_ref_picture_id_;
}

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_RPS_TEST_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_RPS_TEST_H_
#include "vp8.h"
#include "normal_async_test.h"
class RpsDecodeCompleteCallback;
class VP8RpsTest : public VP8NormalAsyncTest {
public:
VP8RpsTest(float bitRate);
VP8RpsTest();
virtual ~VP8RpsTest();
virtual void Perform();
private:
VP8RpsTest(std::string name, std::string description, unsigned int testNo)
: VP8NormalAsyncTest(name, description, testNo) {}
virtual bool EncodeRps(RpsDecodeCompleteCallback* decodeCallback);
virtual int Decode(int lossValue = 0);
static bool CheckIfBitExact(const void *ptrA, unsigned int aLengthBytes,
const void *ptrB, unsigned int bLengthBytes);
webrtc::VP8Decoder* decoder2_;
webrtc::VideoFrame decoded_frame2_;
bool sli_;
};
class RpsDecodeCompleteCallback : public webrtc::DecodedImageCallback {
public:
RpsDecodeCompleteCallback(webrtc::VideoFrame* buffer);
WebRtc_Word32 Decoded(webrtc::VideoFrame& decodedImage);
bool DecodeComplete();
WebRtc_Word32 ReceivedDecodedReferenceFrame(const WebRtc_UWord64 picture_id);
WebRtc_Word32 ReceivedDecodedFrame(const WebRtc_UWord64 picture_id);
WebRtc_UWord64 LastDecodedPictureId() const;
WebRtc_UWord64 LastDecodedRefPictureId(bool *updated);
private:
webrtc::VideoFrame* decoded_frame_;
bool decode_complete_;
WebRtc_UWord64 last_decoded_picture_id_;
WebRtc_UWord64 last_decoded_ref_picture_id_;
bool updated_ref_picture_id_;
};
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_RPS_TEST_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <fstream>
#include <iostream>
#include <vector>
#include "benchmark.h"
#include "dual_decoder_test.h"
#include "gtest/gtest.h"
#include "normal_async_test.h"
#include "packet_loss_test.h"
#include "vp8_unittest.h"
#include "rps_test.h"
#include "testsupport/fileutils.h"
#include "vp8.h"
using namespace webrtc;
void PopulateTests(std::vector<CodecTest*>* tests)
{
// tests->push_back(new VP8RpsTest());
tests->push_back(new VP8UnitTest());
// tests->push_back(new VP8DualDecoderTest());
// tests->push_back(new VP8Benchmark());
// tests->push_back(new VP8PacketLossTest(0.05, false, 5));
// tests->push_back(new VP8NormalAsyncTest());
}
TEST(Vp8WrapperTest, RunAllTests)
{
VP8Encoder* enc;
VP8Decoder* dec;
std::vector<CodecTest*> tests;
PopulateTests(&tests);
std::fstream log;
std::string log_file = webrtc::test::OutputPath() + "VP8_test_log.txt";
log.open(log_file.c_str(), std::fstream::out | std::fstream::app);
std::vector<CodecTest*>::iterator it;
for (it = tests.begin() ; it < tests.end(); it++)
{
enc = VP8Encoder::Create();
dec = VP8Decoder::Create();
(*it)->SetEncoder(enc);
(*it)->SetDecoder(dec);
(*it)->SetLog(&log);
(*it)->Perform();
(*it)->Print();
delete enc;
delete dec;
delete *it;
}
log.close();
tests.pop_back();
}

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "vp8_unittest.h"
#include <string.h>
#include "modules/video_coding/codecs/test_framework/video_source.h"
#include "gtest/gtest.h"
#include "testsupport/fileutils.h"
#include "vp8.h"
using namespace webrtc;
VP8UnitTest::VP8UnitTest()
:
UnitTest("VP8UnitTest", "Unit test")
{
}
VP8UnitTest::VP8UnitTest(std::string name, std::string description)
:
UnitTest(name, description)
{
}
WebRtc_UWord32
VP8UnitTest::CodecSpecific_SetBitrate(WebRtc_UWord32 bitRate,
WebRtc_UWord32 /*frameRate*/)
{
int rate = _encoder->SetRates(bitRate, _inst.maxFramerate);
EXPECT_TRUE(rate >= 0);
return rate;
}
void
VP8UnitTest::Perform()
{
Setup();
VP8Encoder* enc = (VP8Encoder*)_encoder;
VP8Decoder* dec = (VP8Decoder*)_decoder;
//----- Encoder parameter tests -----
//-- Calls before InitEncode() --
EXPECT_EQ(enc->Release(), WEBRTC_VIDEO_CODEC_OK);
EXPECT_EQ(enc->SetRates(_bitRate, _inst.maxFramerate),
WEBRTC_VIDEO_CODEC_UNINITIALIZED);
EXPECT_EQ(enc->SetRates(_bitRate, _inst.maxFramerate),
WEBRTC_VIDEO_CODEC_UNINITIALIZED);
VideoCodec codecInst;
memset(&codecInst, 0, sizeof(codecInst));
strncpy(codecInst.plName, "VP8", 31);
codecInst.plType = 126;
codecInst.maxBitrate = 0;
codecInst.minBitrate = 0;
codecInst.width = 1440;
codecInst.height = 1080;
codecInst.maxFramerate = 30;
codecInst.startBitrate = 300;
codecInst.codecSpecific.VP8.complexity = kComplexityNormal;
codecInst.codecSpecific.VP8.numberOfTemporalLayers = 1;
EXPECT_EQ(enc->InitEncode(&codecInst, 1, 1440), WEBRTC_VIDEO_CODEC_OK);
//-- Test two problematic level settings --
strncpy(codecInst.plName, "VP8", 31);
codecInst.plType = 126;
codecInst.maxBitrate = 0;
codecInst.minBitrate = 0;
codecInst.width = 352;
codecInst.height = 288;
codecInst.maxFramerate = 30;
codecInst.codecSpecific.VP8.complexity = kComplexityNormal;
codecInst.startBitrate = 300;
EXPECT_EQ(enc->InitEncode(&codecInst, 1, 1440), WEBRTC_VIDEO_CODEC_OK);
// Settings not correct for this profile
strncpy(codecInst.plName, "VP8", 31);
codecInst.plType = 126;
codecInst.maxBitrate = 0;
codecInst.minBitrate = 0;
codecInst.width = 176;
codecInst.height = 144;
codecInst.maxFramerate = 15;
codecInst.codecSpecific.VP8.complexity = kComplexityNormal;
codecInst.startBitrate = 300;
ASSERT_EQ(enc->InitEncode(&_inst, 1, 1440), WEBRTC_VIDEO_CODEC_OK);
//-- ProcessNewBitrate() errors --
// Bad bitrate.
EXPECT_EQ(enc->SetRates(_inst.maxBitrate + 1, _inst.maxFramerate),
WEBRTC_VIDEO_CODEC_OK);
//----- Decoder parameter tests -----
//-- Calls before InitDecode() --
EXPECT_TRUE(dec->Release() == 0);
ASSERT_TRUE(dec->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
//-- SetCodecConfigParameters() errors --
unsigned char tmpBuf[128];
EXPECT_TRUE(dec->SetCodecConfigParameters(NULL, sizeof(tmpBuf)) == -1);
EXPECT_TRUE(dec->SetCodecConfigParameters(tmpBuf, 1) == -1);
// Garbage data.
EXPECT_TRUE(dec->SetCodecConfigParameters(tmpBuf, sizeof(tmpBuf)) == -1);
UnitTest::Perform();
Teardown();
}

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_VP8_UNITTEST_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_VP8_UNITTEST_H_
#include "modules/video_coding/codecs/test_framework/unit_test.h"
class VP8UnitTest : public UnitTest
{
public:
VP8UnitTest();
VP8UnitTest(std::string name, std::string description);
virtual void Perform();
protected:
virtual WebRtc_UWord32 CodecSpecific_SetBitrate(
WebRtc_UWord32 bitRate,
WebRtc_UWord32 /*frameRate*/);
};
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_VP8_UNITTEST_H_

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# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': [
'../../../../build/common.gypi',
'../test_framework/test_framework.gypi'
],
'targets': [
{
'target_name': 'webrtc_vp8',
'type': '<(library)',
'dependencies': [
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/common_video/common_video.gyp:common_video',
],
'include_dirs': [
'include',
'<(webrtc_root)/common_video/interface',
'<(webrtc_root)/modules/video_coding/codecs/interface',
'<(webrtc_root)/modules/interface',
],
'conditions': [
['build_libvpx==1', {
'dependencies': [
'<(DEPTH)/third_party/libvpx/libvpx.gyp:libvpx',
],
}],
# TODO(mikhal): Investigate this mechanism for handling differences
# between the Chromium and standalone builds.
# http://code.google.com/p/webrtc/issues/detail?id=201
['build_with_chromium==1', {
'defines': [
'WEBRTC_LIBVPX_VERSION=960' # Bali
],
}, {
'defines': [
'WEBRTC_LIBVPX_VERSION=971' # Cayuga
],
'sources': [
'temporal_layers.h',
'temporal_layers.cc',
],
}],
],
'direct_dependent_settings': {
'include_dirs': [
'include',
'<(webrtc_root)/common_video/interface',
'<(webrtc_root)/modules/video_coding/codecs/interface',
],
},
'sources': [
'reference_picture_selection.h',
'reference_picture_selection.cc',
'include/vp8.h',
'include/vp8_common_types.h',
'vp8_impl.cc',
],
},
], # targets
'conditions': [
['include_tests==1', {
'targets': [
{
'target_name': 'vp8_integrationtests',
'type': 'executable',
'dependencies': [
'test_framework',
'webrtc_vp8',
'<(webrtc_root)/common_video/common_video.gyp:common_video',
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/test/test.gyp:test_support',
'<(webrtc_root)/test/test.gyp:test_support_main',
'<(DEPTH)/testing/gtest.gyp:gtest',
],
'sources': [
# header files
'test/benchmark.h',
'test/dual_decoder_test.h',
'test/normal_async_test.h',
'test/packet_loss_test.h',
'test/rps_test.h',
'test/vp8_unittest.h',
# source files
'test/benchmark.cc',
'test/dual_decoder_test.cc',
'test/normal_async_test.cc',
'test/packet_loss_test.cc',
'test/rps_test.cc',
'test/tester.cc',
'test/vp8_unittest.cc',
],
},
{
'target_name': 'vp8_unittests',
'type': 'executable',
'dependencies': [
'webrtc_vp8',
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/test/test.gyp:test_support_main',
],
'include_dirs': [
'<(DEPTH)/third_party/libvpx/source/libvpx',
],
'sources': [
'reference_picture_selection_unittest.cc',
'temporal_layers_unittest.cc',
],
'conditions': [
['build_libvpx==1', {
'dependencies': [
'<(DEPTH)/third_party/libvpx/libvpx.gyp:libvpx',
],
}],
],
},
], # targets
}], # include_tests
],
}

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*
* WEBRTC VP8 wrapper interface
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_IMPL_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_IMPL_H_
#include "modules/video_coding/codecs/vp8/include/vp8.h"
// VPX forward declaration
typedef struct vpx_codec_ctx vpx_codec_ctx_t;
typedef struct vpx_codec_ctx vpx_dec_ctx_t;
typedef struct vpx_codec_enc_cfg vpx_codec_enc_cfg_t;
typedef struct vpx_image vpx_image_t;
typedef struct vpx_ref_frame vpx_ref_frame_t;
struct vpx_codec_cx_pkt;
namespace webrtc {
class TemporalLayers;
class ReferencePictureSelection;
class VP8EncoderImpl : public VP8Encoder {
public:
VP8EncoderImpl();
virtual ~VP8EncoderImpl();
// Free encoder memory.
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
virtual int Release();
// Initialize the encoder with the information from the codecSettings
//
// Input:
// - codec_settings : Codec settings
// - number_of_cores : Number of cores available for the encoder
// - max_payload_size : The maximum size each payload is allowed
// to have. Usually MTU - overhead.
//
// Return value : Set bit rate if OK
// <0 - Errors:
// WEBRTC_VIDEO_CODEC_ERR_PARAMETER
// WEBRTC_VIDEO_CODEC_ERR_SIZE
// WEBRTC_VIDEO_CODEC_LEVEL_EXCEEDED
// WEBRTC_VIDEO_CODEC_MEMORY
// WEBRTC_VIDEO_CODEC_ERROR
virtual int InitEncode(const VideoCodec* codec_settings,
int number_of_cores,
uint32_t max_payload_size);
// Encode an I420 image (as a part of a video stream). The encoded image
// will be returned to the user through the encode complete callback.
//
// Input:
// - input_image : Image to be encoded
// - frame_types : Frame type to be generated by the encoder.
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK
// <0 - Errors:
// WEBRTC_VIDEO_CODEC_ERR_PARAMETER
// WEBRTC_VIDEO_CODEC_MEMORY
// WEBRTC_VIDEO_CODEC_ERROR
// WEBRTC_VIDEO_CODEC_TIMEOUT
virtual int Encode(const VideoFrame& input_image,
const CodecSpecificInfo* codec_specific_info,
const std::vector<VideoFrameType>* frame_types);
// Register an encode complete callback object.
//
// Input:
// - callback : Callback object which handles encoded images.
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
virtual int RegisterEncodeCompleteCallback(EncodedImageCallback* callback);
// Inform the encoder of the new packet loss rate and the round-trip time of
// the network.
//
// - packet_loss : Fraction lost
// (loss rate in percent = 100 * packetLoss / 255)
// - rtt : Round-trip time in milliseconds
// Return value : WEBRTC_VIDEO_CODEC_OK if OK
// <0 - Errors: WEBRTC_VIDEO_CODEC_ERROR
//
virtual int SetChannelParameters(uint32_t packet_loss, int rtt);
// Inform the encoder about the new target bit rate.
//
// - new_bitrate_kbit : New target bit rate
// - frame_rate : The target frame rate
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
virtual int SetRates(uint32_t new_bitrate_kbit, uint32_t frame_rate);
private:
// Call encoder initialize function and set control settings.
int InitAndSetControlSettings(const VideoCodec* inst);
// Update frame size for codec.
int UpdateCodecFrameSize(WebRtc_UWord32 input_image_width,
WebRtc_UWord32 input_image_height);
void PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
const vpx_codec_cx_pkt& pkt,
uint32_t timestamp);
int GetEncodedFrame(const VideoFrame& input_image);
int GetEncodedPartitions(const VideoFrame& input_image);
// Determine maximum target for Intra frames
//
// Input:
// - optimal_buffer_size : Optimal buffer size
// Return Value : Max target size for Intra frames represented as
// percentage of the per frame bandwidth
uint32_t MaxIntraTarget(uint32_t optimal_buffer_size);
EncodedImage encoded_image_;
EncodedImageCallback* encoded_complete_callback_;
VideoCodec codec_;
bool inited_;
int64_t timestamp_;
uint16_t picture_id_;
bool feedback_mode_;
int cpu_speed_;
uint32_t rc_max_intra_target_;
int token_partitions_;
ReferencePictureSelection* rps_;
TemporalLayers* temporal_layers_;
vpx_codec_ctx_t* encoder_;
vpx_codec_enc_cfg_t* config_;
vpx_image_t* raw_;
}; // end of VP8Encoder class
class VP8DecoderImpl : public VP8Decoder {
public:
VP8DecoderImpl();
virtual ~VP8DecoderImpl();
// Initialize the decoder.
//
// Return value : WEBRTC_VIDEO_CODEC_OK.
// <0 - Errors:
// WEBRTC_VIDEO_CODEC_ERROR
virtual int InitDecode(const VideoCodec* inst, int number_of_cores);
// Decode encoded image (as a part of a video stream). The decoded image
// will be returned to the user through the decode complete callback.
//
// Input:
// - input_image : Encoded image to be decoded
// - missing_frames : True if one or more frames have been lost
// since the previous decode call.
// - fragmentation : Specifies the start and length of each VP8
// partition.
// - codec_specific_info : pointer to specific codec data
// - render_time_ms : Render time in Ms
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK
// <0 - Errors:
// WEBRTC_VIDEO_CODEC_ERROR
// WEBRTC_VIDEO_CODEC_ERR_PARAMETER
virtual int Decode(const EncodedImage& input_image,
bool missing_frames,
const RTPFragmentationHeader* fragmentation,
const CodecSpecificInfo* codec_specific_info,
int64_t /*render_time_ms*/);
// Register a decode complete callback object.
//
// Input:
// - callback : Callback object which handles decoded images.
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
virtual int RegisterDecodeCompleteCallback(DecodedImageCallback* callback);
// Free decoder memory.
//
// Return value : WEBRTC_VIDEO_CODEC_OK if OK
// <0 - Errors:
// WEBRTC_VIDEO_CODEC_ERROR
virtual int Release();
// Reset decoder state and prepare for a new call.
//
// Return value : WEBRTC_VIDEO_CODEC_OK.
// <0 - Errors:
// WEBRTC_VIDEO_CODEC_UNINITIALIZED
// WEBRTC_VIDEO_CODEC_ERROR
virtual int Reset();
// Create a copy of the codec and its internal state.
//
// Return value : A copy of the instance if OK, NULL otherwise.
virtual VideoDecoder* Copy();
private:
// Copy reference image from this _decoder to the _decoder in copyTo. Set
// which frame type to copy in _refFrame->frame_type before the call to
// this function.
int CopyReference(VP8Decoder* copy);
int DecodePartitions(const EncodedImage& input_image,
const RTPFragmentationHeader* fragmentation);
int ReturnFrame(const vpx_image_t* img, uint32_t timeStamp);
VideoFrame decoded_image_;
DecodedImageCallback* decode_complete_callback_;
bool inited_;
bool feedback_mode_;
vpx_dec_ctx_t* decoder_;
VideoCodec codec_;
EncodedImage last_keyframe_;
int image_format_;
vpx_ref_frame_t* ref_frame_;
int propagation_cnt_;
bool latest_keyframe_complete_;
bool mfqe_enabled_;
}; // end of VP8Decoder class
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_IMPL_H_

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stefan@webrtc.org
mikhal@webrtc.org
marpan@webrtc.org
henrik.lundin@webrtc.org

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_INTERFACE_MOCK_MOCK_VCM_CALLBACKS_H_
#define WEBRTC_MODULES_VIDEO_CODING_MAIN_INTERFACE_MOCK_MOCK_VCM_CALLBACKS_H_
#include "gmock/gmock.h"
#include "typedefs.h"
namespace webrtc {
class MockVCMFrameTypeCallback : public VCMFrameTypeCallback {
public:
MOCK_METHOD0(RequestKeyFrame, int32_t());
MOCK_METHOD1(SliceLossIndicationRequest,
WebRtc_Word32(const WebRtc_UWord64 pictureId));
};
class MockPacketRequestCallback : public VCMPacketRequestCallback {
public:
MOCK_METHOD2(ResendPackets, int32_t(const uint16_t* sequenceNumbers,
uint16_t length));
};
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_MAIN_INTERFACE_MOCK_MOCK_VCM_CALLBACKS_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_INTERFACE_VIDEO_CODING_H_
#define WEBRTC_MODULES_INTERFACE_VIDEO_CODING_H_
#include "modules/interface/module.h"
#include "modules/interface/module_common_types.h"
#include "modules/video_coding/main/interface/video_coding_defines.h"
namespace webrtc
{
class TickTimeBase;
class VideoEncoder;
class VideoDecoder;
struct CodecSpecificInfo;
class VideoCodingModule : public Module
{
public:
enum SenderNackMode {
kNackNone,
kNackAll,
kNackSelective
};
enum ReceiverRobustness {
kNone,
kHardNack,
kSoftNack,
kDualDecoder,
kReferenceSelection
};
enum DecodeErrors {
kNoDecodeErrors,
kAllowDecodeErrors
};
static VideoCodingModule* Create(const WebRtc_Word32 id);
static VideoCodingModule* Create(const WebRtc_Word32 id,
TickTimeBase* clock);
static void Destroy(VideoCodingModule* module);
// Get number of supported codecs
//
// Return value : Number of supported codecs
static WebRtc_UWord8 NumberOfCodecs();
// Get supported codec settings with using id
//
// Input:
// - listId : Id or index of the codec to look up
// - codec : Memory where the codec settings will be stored
//
// Return value : VCM_OK, on success
// VCM_PARAMETER_ERROR if codec not supported or id too high
static WebRtc_Word32 Codec(const WebRtc_UWord8 listId, VideoCodec* codec);
// Get supported codec settings using codec type
//
// Input:
// - codecType : The codec type to get settings for
// - codec : Memory where the codec settings will be stored
//
// Return value : VCM_OK, on success
// VCM_PARAMETER_ERROR if codec not supported
static WebRtc_Word32 Codec(VideoCodecType codecType, VideoCodec* codec);
/*
* Sender
*/
// Any encoder-related state of VCM will be initialized to the
// same state as when the VCM was created. This will not interrupt
// or effect decoding functionality of VCM. VCM will lose all the
// encoding-related settings by calling this function.
// For instance, a send codec has to be registered again.
//
// Return value : VCM_OK, on success.
// < 0, on error.
virtual WebRtc_Word32 InitializeSender() = 0;
// Registers a codec to be used for encoding. Calling this
// API multiple times overwrites any previously registered codecs.
//
// Input:
// - sendCodec : Settings for the codec to be registered.
// - numberOfCores : The number of cores the codec is allowed
// to use.
// - maxPayloadSize : The maximum size each payload is allowed
// to have. Usually MTU - overhead.
//
// Return value : VCM_OK, on success.
// < 0, on error.
virtual WebRtc_Word32 RegisterSendCodec(const VideoCodec* sendCodec,
WebRtc_UWord32 numberOfCores,
WebRtc_UWord32 maxPayloadSize) = 0;
// API to get the current send codec in use.
//
// Input:
// - currentSendCodec : Address where the sendCodec will be written.
//
// Return value : VCM_OK, on success.
// < 0, on error.
virtual WebRtc_Word32 SendCodec(VideoCodec* currentSendCodec) const = 0;
// API to get the current send codec type
//
// Return value : Codec type, on success.
// kVideoCodecUnknown, on error or if no send codec is set
virtual VideoCodecType SendCodec() const = 0;
// Register an external encoder object. This can not be used together with
// external decoder callbacks.
//
// Input:
// - externalEncoder : Encoder object to be used for encoding frames inserted
// with the AddVideoFrame API.
// - payloadType : The payload type bound which this encoder is bound to.
//
// Return value : VCM_OK, on success.
// < 0, on error.
virtual WebRtc_Word32 RegisterExternalEncoder(VideoEncoder* externalEncoder,
WebRtc_UWord8 payloadType,
bool internalSource = false) = 0;
// API to get codec config parameters to be sent out-of-band to a receiver.
//
// Input:
// - buffer : Memory where the codec config parameters should be written.
// - size : Size of the memory available.
//
// Return value : Number of bytes written, on success.
// < 0, on error.
virtual WebRtc_Word32 CodecConfigParameters(WebRtc_UWord8* buffer, WebRtc_Word32 size) = 0;
// API to get currently configured encoder target bitrate in kbit/s.
//
// Return value : 0, on success.
// < 0, on error.
virtual int Bitrate(unsigned int* bitrate) const = 0;
// API to get currently configured encoder target frame rate.
//
// Return value : 0, on success.
// < 0, on error.
virtual int FrameRate(unsigned int* framerate) const = 0;
// Sets the parameters describing the send channel. These parameters are inputs to the
// Media Optimization inside the VCM and also specifies the target bit rate for the
// encoder. Bit rate used by NACK should already be compensated for by the user.
//
// Input:
// - availableBandWidth : Band width available for the VCM in kbit/s.
// - lossRate : Fractions of lost packets the past second.
// (loss rate in percent = 100 * packetLoss / 255)
// - rtt : Current round-trip time in ms.
//
// Return value : VCM_OK, on success.
// < 0, on error.
virtual WebRtc_Word32 SetChannelParameters(WebRtc_UWord32 availableBandWidth,
WebRtc_UWord8 lossRate,
WebRtc_UWord32 rtt) = 0;
// Sets the parameters describing the receive channel. These parameters are inputs to the
// Media Optimization inside the VCM.
//
// Input:
// - rtt : Current round-trip time in ms.
// with the most amount available bandwidth in a conference
// scenario
//
// Return value : VCM_OK, on success.
// < 0, on error.
virtual WebRtc_Word32 SetReceiveChannelParameters(WebRtc_UWord32 rtt) = 0;
// Register a transport callback which will be called to deliver the encoded data and
// side information.
//
// Input:
// - transport : The callback object to register.
//
// Return value : VCM_OK, on success.
// < 0, on error.
virtual WebRtc_Word32 RegisterTransportCallback(VCMPacketizationCallback* transport) = 0;
// Register video output information callback which will be called to deliver information
// about the video stream produced by the encoder, for instance the average frame rate and
// bit rate.
//
// Input:
// - outputInformation : The callback object to register.
//
// Return value : VCM_OK, on success.
// < 0, on error.
virtual WebRtc_Word32 RegisterSendStatisticsCallback(
VCMSendStatisticsCallback* sendStats) = 0;
// Register a video quality settings callback which will be called when
// frame rate/dimensions need to be updated for video quality optimization
//
// Input:
// - videoQMSettings : The callback object to register.
//
// Return value : VCM_OK, on success.
// < 0, on error
virtual WebRtc_Word32 RegisterVideoQMCallback(VCMQMSettingsCallback* videoQMSettings) = 0;
// Register a video protection callback which will be called to deliver
// the requested FEC rate and NACK status (on/off).
//
// Input:
// - protection : The callback object to register.
//
// Return value : VCM_OK, on success.
// < 0, on error.
virtual WebRtc_Word32 RegisterProtectionCallback(VCMProtectionCallback* protection) = 0;
// Enable or disable a video protection method.
//
// Input:
// - videoProtection : The method to enable or disable.
// - enable : True if the method should be enabled, false if
// it should be disabled.
//
// Return value : VCM_OK, on success.
// < 0, on error.
virtual WebRtc_Word32 SetVideoProtection(VCMVideoProtection videoProtection,
bool enable) = 0;
// Add one raw video frame to the encoder. This function does all the necessary
// processing, then decides what frame type to encode, or if the frame should be
// dropped. If the frame should be encoded it passes the frame to the encoder
// before it returns.
//
// Input:
// - videoFrame : Video frame to encode.
// - codecSpecificInfo : Extra codec information, e.g., pre-parsed in-band signaling.
//
// Return value : VCM_OK, on success.
// < 0, on error.
virtual WebRtc_Word32 AddVideoFrame(
const VideoFrame& videoFrame,
const VideoContentMetrics* contentMetrics = NULL,
const CodecSpecificInfo* codecSpecificInfo = NULL) = 0;
// Next frame encoded should be an intra frame (keyframe).
//
// Return value : VCM_OK, on success.
// < 0, on error.
virtual WebRtc_Word32 IntraFrameRequest(int stream_index) = 0;
// Frame Dropper enable. Can be used to disable the frame dropping when the encoder
// over-uses its bit rate. This API is designed to be used when the encoded frames
// are supposed to be stored to an AVI file, or when the I420 codec is used and the
// target bit rate shouldn't affect the frame rate.
//
// Input:
// - enable : True to enable the setting, false to disable it.
//
// Return value : VCM_OK, on success.
// < 0, on error.
virtual WebRtc_Word32 EnableFrameDropper(bool enable) = 0;
// Sent frame counters
virtual WebRtc_Word32 SentFrameCount(VCMFrameCount& frameCount) const = 0;
/*
* Receiver
*/
// The receiver state of the VCM will be initialized to the
// same state as when the VCM was created. This will not interrupt
// or effect the send side functionality of VCM. VCM will lose all the
// decoding-related settings by calling this function. All frames
// inside the jitter buffer are flushed and the delay is reset.
// For instance, a receive codec has to be registered again.
//
// Return value : VCM_OK, on success.
// < 0, on error.
virtual WebRtc_Word32 InitializeReceiver() = 0;
// Register possible receive codecs, can be called multiple times for different codecs.
// The module will automatically switch between registered codecs depending on the
// payload type of incoming frames. The actual decoder will be created when needed.
//
// Input:
// - receiveCodec : Settings for the codec to be registered.
// - numberOfCores : Number of CPU cores that the decoder is allowed to use.
// - requireKeyFrame : Set this to true if you don't want any delta frames
// to be decoded until the first key frame has been decoded.
//
// Return value : VCM_OK, on success.
// < 0, on error.
virtual WebRtc_Word32 RegisterReceiveCodec(const VideoCodec* receiveCodec,
WebRtc_Word32 numberOfCores,
bool requireKeyFrame = false) = 0;
// Register an externally defined decoder/renderer object. Can be a decoder only or a
// decoder coupled with a renderer. Note that RegisterReceiveCodec must be called to
// be used for decoding incoming streams.
//
// Input:
// - externalDecoder : The external decoder/renderer object.
// - payloadType : The payload type which this decoder should be
// registered to.
// - internalRenderTiming : True if the internal renderer (if any) of the decoder
// object can make sure to render at a given time in ms.
//
// Return value : VCM_OK, on success.
// < 0, on error.
virtual WebRtc_Word32 RegisterExternalDecoder(VideoDecoder* externalDecoder,
WebRtc_UWord8 payloadType,
bool internalRenderTiming) = 0;
// Register a receive callback. Will be called whenever there is a new frame ready
// for rendering.
//
// Input:
// - receiveCallback : The callback object to be used by the module when a
// frame is ready for rendering.
// De-register with a NULL pointer.
//
// Return value : VCM_OK, on success.
// < 0, on error.
virtual WebRtc_Word32 RegisterReceiveCallback(VCMReceiveCallback* receiveCallback) = 0;
// Register a receive statistics callback which will be called to deliver information
// about the video stream received by the receiving side of the VCM, for instance the
// average frame rate and bit rate.
//
// Input:
// - receiveStats : The callback object to register.
//
// Return value : VCM_OK, on success.
// < 0, on error.
virtual WebRtc_Word32 RegisterReceiveStatisticsCallback(
VCMReceiveStatisticsCallback* receiveStats) = 0;
// Register a frame type request callback. This callback will be called when the
// module needs to request specific frame types from the send side.
//
// Input:
// - frameTypeCallback : The callback object to be used by the module when
// requesting a specific type of frame from the send side.
// De-register with a NULL pointer.
//
// Return value : VCM_OK, on success.
// < 0, on error.
virtual WebRtc_Word32 RegisterFrameTypeCallback(
VCMFrameTypeCallback* frameTypeCallback) = 0;
// Register a frame storage callback. This callback will be called right before an
// encoded frame is given to the decoder. Useful for recording the incoming video sequence.
//
// Input:
// - frameStorageCallback : The callback object used by the module
// to store a received encoded frame.
//
// Return value : VCM_OK, on success.
// < 0, on error.
virtual WebRtc_Word32 RegisterFrameStorageCallback(
VCMFrameStorageCallback* frameStorageCallback) = 0;
// Registers a callback which is called whenever the receive side of the VCM
// encounters holes in the packet sequence and needs packets to be retransmitted.
//
// Input:
// - callback : The callback to be registered in the VCM.
//
// Return value : VCM_OK, on success.
// <0, on error.
virtual WebRtc_Word32 RegisterPacketRequestCallback(
VCMPacketRequestCallback* callback) = 0;
// Waits for the next frame in the jitter buffer to become complete
// (waits no longer than maxWaitTimeMs), then passes it to the decoder for decoding.
// Should be called as often as possible to get the most out of the decoder.
//
// Return value : VCM_OK, on success.
// < 0, on error.
virtual WebRtc_Word32 Decode(WebRtc_UWord16 maxWaitTimeMs = 200) = 0;
// Waits for the next frame in the dual jitter buffer to become complete
// (waits no longer than maxWaitTimeMs), then passes it to the dual decoder
// for decoding. This will never trigger a render callback. Should be
// called frequently, and as long as it returns 1 it should be called again
// as soon as possible.
//
// Return value : 1, if a frame was decoded
// 0, if no frame was decoded
// < 0, on error.
virtual WebRtc_Word32 DecodeDualFrame(WebRtc_UWord16 maxWaitTimeMs = 200) = 0;
// Decodes a frame and sets an appropriate render time in ms relative to the system time.
// Should be used in conjunction with VCMFrameStorageCallback.
//
// Input:
// - frameFromStorage : Encoded frame read from file or received through
// the VCMFrameStorageCallback callback.
//
// Return value: : VCM_OK, on success
// < 0, on error
virtual WebRtc_Word32 DecodeFromStorage(const EncodedVideoData& frameFromStorage) = 0;
// Reset the decoder state to the initial state.
//
// Return value : VCM_OK, on success.
// < 0, on error.
virtual WebRtc_Word32 ResetDecoder() = 0;
// API to get the codec which is currently used for decoding by the module.
//
// Input:
// - currentReceiveCodec : Settings for the codec to be registered.
//
// Return value : VCM_OK, on success.
// < 0, on error.
virtual WebRtc_Word32 ReceiveCodec(VideoCodec* currentReceiveCodec) const = 0;
// API to get the codec type currently used for decoding by the module.
//
// Return value : codecy type, on success.
// kVideoCodecUnknown, on error or if no receive codec is registered
virtual VideoCodecType ReceiveCodec() const = 0;
// Insert a parsed packet into the receiver side of the module. Will be placed in the
// jitter buffer waiting for the frame to become complete. Returns as soon as the packet
// has been placed in the jitter buffer.
//
// Input:
// - incomingPayload : Payload of the packet.
// - payloadLength : Length of the payload.
// - rtpInfo : The parsed header.
//
// Return value : VCM_OK, on success.
// < 0, on error.
virtual WebRtc_Word32 IncomingPacket(const WebRtc_UWord8* incomingPayload,
WebRtc_UWord32 payloadLength,
const WebRtcRTPHeader& rtpInfo) = 0;
// Minimum playout delay (Used for lip-sync). This is the minimum delay required
// to sync with audio. Not included in VideoCodingModule::Delay()
// Defaults to 0 ms.
//
// Input:
// - minPlayoutDelayMs : Additional delay in ms.
//
// Return value : VCM_OK, on success.
// < 0, on error.
virtual WebRtc_Word32 SetMinimumPlayoutDelay(WebRtc_UWord32 minPlayoutDelayMs) = 0;
// Set the time required by the renderer to render a frame.
//
// Input:
// - timeMS : The time in ms required by the renderer to render a frame.
//
// Return value : VCM_OK, on success.
// < 0, on error.
virtual WebRtc_Word32 SetRenderDelay(WebRtc_UWord32 timeMS) = 0;
// The total delay desired by the VCM. Can be less than the minimum
// delay set with SetMinimumPlayoutDelay.
//
// Return value : Total delay in ms, on success.
// < 0, on error.
virtual WebRtc_Word32 Delay() const = 0;
// Get the received frame counters. Keeps track of the number of each frame type
// received since the start of the call.
//
// Output:
// - frameCount : Struct to be filled with the number of frames received.
//
// Return value : VCM_OK, on success.
// <0, on error.
virtual WebRtc_Word32 ReceivedFrameCount(VCMFrameCount& frameCount) const = 0;
// Returns the number of packets discarded by the jitter buffer due to being
// too late. This can include duplicated packets which arrived after the
// frame was sent to the decoder. Therefore packets which were prematurely
// NACKed will be counted.
virtual WebRtc_UWord32 DiscardedPackets() const = 0;
// Robustness APIs
// Set the sender RTX/NACK mode.
// Input:
// - mode : the selected NACK mode.
//
// Return value : VCM_OK, on success;
// < 0, on error.
virtual int SetSenderNackMode(SenderNackMode mode) = 0;
// Set the sender reference picture selection (RPS) mode.
// Input:
// - enable : true or false, for enable and disable, respectively.
//
// Return value : VCM_OK, on success;
// < 0, on error.
virtual int SetSenderReferenceSelection(bool enable) = 0;
// Set the sender forward error correction (FEC) mode.
// Input:
// - enable : true or false, for enable and disable, respectively.
//
// Return value : VCM_OK, on success;
// < 0, on error.
virtual int SetSenderFEC(bool enable) = 0;
// Set the key frame period, or disable periodic key frames (I-frames).
// Input:
// - periodMs : period in ms; <= 0 to disable periodic key frames.
//
// Return value : VCM_OK, on success;
// < 0, on error.
virtual int SetSenderKeyFramePeriod(int periodMs) = 0;
// Set the receiver robustness mode. The mode decides how the receiver
// responds to losses in the stream. The type of counter-measure (soft or
// hard NACK, dual decoder, RPS, etc.) is selected through the
// robustnessMode parameter. The errorMode parameter decides if it is
// allowed to display frames corrupted by losses. Note that not all
// combinations of the two parameters are feasible. An error will be
// returned for invalid combinations.
// Input:
// - robustnessMode : selected robustness mode.
// - errorMode : selected error mode.
//
// Return value : VCM_OK, on success;
// < 0, on error.
virtual int SetReceiverRobustnessMode(ReceiverRobustness robustnessMode,
DecodeErrors errorMode) = 0;
// Enables recording of debugging information.
virtual int StartDebugRecording(const char* file_name_utf8) = 0;
// Disables recording of debugging information.
virtual int StopDebugRecording() = 0;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_INTERFACE_VIDEO_CODING_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_INTERFACE_VIDEO_CODING_DEFINES_H_
#define WEBRTC_MODULES_INTERFACE_VIDEO_CODING_DEFINES_H_
#include "typedefs.h"
#include "modules/interface/module_common_types.h"
namespace webrtc {
// Error codes
#define VCM_FRAME_NOT_READY 3
#define VCM_REQUEST_SLI 2
#define VCM_MISSING_CALLBACK 1
#define VCM_OK 0
#define VCM_GENERAL_ERROR -1
#define VCM_LEVEL_EXCEEDED -2
#define VCM_MEMORY -3
#define VCM_PARAMETER_ERROR -4
#define VCM_UNKNOWN_PAYLOAD -5
#define VCM_CODEC_ERROR -6
#define VCM_UNINITIALIZED -7
#define VCM_NO_CODEC_REGISTERED -8
#define VCM_JITTER_BUFFER_ERROR -9
#define VCM_OLD_PACKET_ERROR -10
#define VCM_NO_FRAME_DECODED -11
#define VCM_ERROR_REQUEST_SLI -12
#define VCM_NOT_IMPLEMENTED -20
#define VCM_RED_PAYLOAD_TYPE 96
#define VCM_ULPFEC_PAYLOAD_TYPE 97
#define VCM_VP8_PAYLOAD_TYPE 120
#define VCM_I420_PAYLOAD_TYPE 124
enum VCMNackProperties {
kNackHistoryLength = 450
};
enum VCMVideoProtection {
kProtectionNack, // Both send-side and receive-side
kProtectionNackSender, // Send-side only
kProtectionNackReceiver, // Receive-side only
kProtectionDualDecoder,
kProtectionFEC,
kProtectionNackFEC,
kProtectionKeyOnLoss,
kProtectionKeyOnKeyLoss,
kProtectionPeriodicKeyFrames
};
enum VCMTemporalDecimation {
kBitrateOverUseDecimation,
};
struct VCMFrameCount {
WebRtc_UWord32 numKeyFrames;
WebRtc_UWord32 numDeltaFrames;
};
// Callback class used for sending data ready to be packetized
class VCMPacketizationCallback {
public:
virtual WebRtc_Word32 SendData(
FrameType frameType,
WebRtc_UWord8 payloadType,
WebRtc_UWord32 timeStamp,
int64_t capture_time_ms,
const WebRtc_UWord8* payloadData,
WebRtc_UWord32 payloadSize,
const RTPFragmentationHeader& fragmentationHeader,
const RTPVideoHeader* rtpVideoHdr) = 0;
protected:
virtual ~VCMPacketizationCallback() {
}
};
// Callback class used for passing decoded frames which are ready to be rendered.
class VCMFrameStorageCallback {
public:
virtual WebRtc_Word32 StoreReceivedFrame(
const EncodedVideoData& frameToStore) = 0;
protected:
virtual ~VCMFrameStorageCallback() {
}
};
// Callback class used for passing decoded frames which are ready to be rendered.
class VCMReceiveCallback {
public:
virtual WebRtc_Word32 FrameToRender(VideoFrame& videoFrame) = 0;
virtual WebRtc_Word32 ReceivedDecodedReferenceFrame(
const WebRtc_UWord64 pictureId) {
return -1;
}
protected:
virtual ~VCMReceiveCallback() {
}
};
// Callback class used for informing the user of the bit rate and frame rate produced by the
// encoder.
class VCMSendStatisticsCallback {
public:
virtual WebRtc_Word32 SendStatistics(const WebRtc_UWord32 bitRate,
const WebRtc_UWord32 frameRate) = 0;
protected:
virtual ~VCMSendStatisticsCallback() {
}
};
// Callback class used for informing the user of the incoming bit rate and frame rate.
class VCMReceiveStatisticsCallback {
public:
virtual WebRtc_Word32 ReceiveStatistics(const WebRtc_UWord32 bitRate,
const WebRtc_UWord32 frameRate) = 0;
protected:
virtual ~VCMReceiveStatisticsCallback() {
}
};
// Callback class used for telling the user about how to configure the FEC,
// and the rates sent the last second is returned to the VCM.
class VCMProtectionCallback {
public:
virtual int ProtectionRequest(const FecProtectionParams* delta_params,
const FecProtectionParams* key_params,
uint32_t* sent_video_rate_bps,
uint32_t* sent_nack_rate_bps,
uint32_t* sent_fec_rate_bps) = 0;
protected:
virtual ~VCMProtectionCallback() {
}
};
// Callback class used for telling the user about what frame type needed to continue decoding.
// Typically a key frame when the stream has been corrupted in some way.
class VCMFrameTypeCallback {
public:
virtual WebRtc_Word32 RequestKeyFrame() = 0;
virtual WebRtc_Word32 SliceLossIndicationRequest(
const WebRtc_UWord64 pictureId) {
return -1;
}
protected:
virtual ~VCMFrameTypeCallback() {
}
};
// Callback class used for telling the user about which packet sequence numbers are currently
// missing and need to be resent.
class VCMPacketRequestCallback {
public:
virtual WebRtc_Word32 ResendPackets(const WebRtc_UWord16* sequenceNumbers,
WebRtc_UWord16 length) = 0;
protected:
virtual ~VCMPacketRequestCallback() {
}
};
// Callback used to inform the user of the the desired resolution
// as subscribed by Media Optimization (Quality Modes)
class VCMQMSettingsCallback {
public:
virtual WebRtc_Word32 SetVideoQMSettings(const WebRtc_UWord32 frameRate,
const WebRtc_UWord32 width,
const WebRtc_UWord32 height) = 0;
protected:
virtual ~VCMQMSettingsCallback() {
}
};
} // namespace webrtc
#endif // WEBRTC_MODULES_INTERFACE_VIDEO_CODING_DEFINES_H_

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# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
LOCAL_PATH := $(call my-dir)
include $(CLEAR_VARS)
include $(LOCAL_PATH)/../../../../../android-webrtc.mk
LOCAL_ARM_MODE := arm
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
LOCAL_MODULE := libwebrtc_video_coding
LOCAL_MODULE_TAGS := optional
LOCAL_CPP_EXTENSION := .cc
LOCAL_SRC_FILES := \
codec_database.cc \
codec_timer.cc \
content_metrics_processing.cc \
decoding_state.cc \
encoded_frame.cc \
exp_filter.cc \
frame_buffer.cc \
frame_dropper.cc \
generic_decoder.cc \
generic_encoder.cc \
inter_frame_delay.cc \
jitter_buffer.cc \
jitter_buffer_common.cc \
jitter_estimator.cc \
media_opt_util.cc \
media_optimization.cc \
packet.cc \
qm_select.cc \
receiver.cc \
rtt_filter.cc \
session_info.cc \
timestamp_extrapolator.cc \
timestamp_map.cc \
timing.cc \
video_coding_impl.cc
# Flags passed to both C and C++ files.
LOCAL_CFLAGS := \
$(MY_WEBRTC_COMMON_DEFS)
LOCAL_C_INCLUDES := \
$(LOCAL_PATH)/../interface \
$(LOCAL_PATH)/../../codecs/interface \
$(LOCAL_PATH)/../../codecs/i420/main/interface \
$(LOCAL_PATH)/../../codecs/vp8/main/interface \
$(LOCAL_PATH)/../../../interface \
$(LOCAL_PATH)/../../../.. \
$(LOCAL_PATH)/../../../../common_video/vplib/main/interface \
$(LOCAL_PATH)/../../../../common_video/interface \
$(LOCAL_PATH)/../../../../system_wrappers/interface
LOCAL_SHARED_LIBRARIES := \
libcutils \
libdl \
libstlport
ifndef NDK_ROOT
include external/stlport/libstlport.mk
endif
include $(BUILD_STATIC_LIBRARY)

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/video_coding/main/source/codec_database.h"
#include <assert.h>
#include "engine_configurations.h"
#ifdef VIDEOCODEC_I420
#include "modules/video_coding/codecs/i420/main/interface/i420.h"
#endif
#ifdef VIDEOCODEC_VP8
#include "modules/video_coding/codecs/vp8/include/vp8.h"
#endif
#include "modules/video_coding/main/source/internal_defines.h"
#include "system_wrappers/interface/trace.h"
namespace webrtc {
VCMDecoderMapItem::VCMDecoderMapItem(VideoCodec* settings,
int number_of_cores,
bool require_key_frame)
: settings(settings),
number_of_cores(number_of_cores),
require_key_frame(require_key_frame) {
assert(number_of_cores >= 0);
}
VCMExtDecoderMapItem::VCMExtDecoderMapItem(
VideoDecoder* external_decoder_instance,
uint8_t payload_type,
bool internal_render_timing)
: payload_type(payload_type),
external_decoder_instance(external_decoder_instance),
internal_render_timing(internal_render_timing) {
}
VCMCodecDataBase::VCMCodecDataBase(int id)
: id_(id),
number_of_cores_(0),
max_payload_size_(kDefaultPayloadSize),
periodic_key_frames_(false),
current_enc_is_external_(false),
send_codec_(),
receive_codec_(),
external_payload_type_(0),
external_encoder_(NULL),
internal_source_(false),
ptr_encoder_(NULL),
ptr_decoder_(NULL),
current_dec_is_external_(false),
dec_map_(),
dec_external_map_() {
}
VCMCodecDataBase::~VCMCodecDataBase() {
ResetSender();
ResetReceiver();
}
int VCMCodecDataBase::NumberOfCodecs() {
return VCM_NUM_VIDEO_CODECS_AVAILABLE;
}
bool VCMCodecDataBase::Codec(int list_id,
VideoCodec* settings) {
if (!settings) {
return false;
}
if (list_id >= VCM_NUM_VIDEO_CODECS_AVAILABLE) {
return false;
}
memset(settings, 0, sizeof(VideoCodec));
switch (list_id) {
#ifdef VIDEOCODEC_VP8
case VCM_VP8_IDX: {
strncpy(settings->plName, "VP8", 4);
settings->codecType = kVideoCodecVP8;
// 96 to 127 dynamic payload types for video codecs.
settings->plType = VCM_VP8_PAYLOAD_TYPE;
settings->startBitrate = 100;
settings->minBitrate = VCM_MIN_BITRATE;
settings->maxBitrate = 0;
settings->maxFramerate = VCM_DEFAULT_FRAME_RATE;
settings->width = VCM_DEFAULT_CODEC_WIDTH;
settings->height = VCM_DEFAULT_CODEC_HEIGHT;
settings->numberOfSimulcastStreams = 0;
settings->codecSpecific.VP8.resilience = kResilientStream;
settings->codecSpecific.VP8.numberOfTemporalLayers = 1;
settings->codecSpecific.VP8.denoisingOn = true;
settings->codecSpecific.VP8.errorConcealmentOn = false;
settings->codecSpecific.VP8.automaticResizeOn = false;
settings->codecSpecific.VP8.frameDroppingOn = true;
return true;
}
#endif
#ifdef VIDEOCODEC_I420
case VCM_I420_IDX: {
strncpy(settings->plName, "I420", 5);
settings->codecType = kVideoCodecI420;
// 96 to 127 dynamic payload types for video codecs.
settings->plType = VCM_I420_PAYLOAD_TYPE;
// Bitrate needed for this size and framerate.
settings->startBitrate = 3 * VCM_DEFAULT_CODEC_WIDTH *
VCM_DEFAULT_CODEC_HEIGHT * 8 *
VCM_DEFAULT_FRAME_RATE / 1000 / 2;
settings->maxBitrate = settings->startBitrate;
settings->maxFramerate = VCM_DEFAULT_FRAME_RATE;
settings->width = VCM_DEFAULT_CODEC_WIDTH;
settings->height = VCM_DEFAULT_CODEC_HEIGHT;
settings->minBitrate = VCM_MIN_BITRATE;
settings->numberOfSimulcastStreams = 0;
return true;
}
#endif
default: {
return false;
}
}
}
bool VCMCodecDataBase::Codec(VideoCodecType codec_type,
VideoCodec* settings) {
for (int i = 0; i < VCMCodecDataBase::NumberOfCodecs(); i++) {
const bool ret = VCMCodecDataBase::Codec(i, settings);
if (!ret) {
return false;
}
if (codec_type == settings->codecType) {
return true;
}
}
return false;
}
void VCMCodecDataBase::ResetSender() {
DeleteEncoder();
periodic_key_frames_ = false;
}
// Assuming only one registered encoder - since only one used, no need for more.
bool VCMCodecDataBase::RegisterSendCodec(
const VideoCodec* send_codec,
int number_of_cores,
int max_payload_size) {
if (!send_codec) {
return false;
}
if (max_payload_size <= 0) {
max_payload_size = kDefaultPayloadSize;
}
if (number_of_cores < 0 || number_of_cores > 32) {
return false;
}
if (send_codec->plType <= 0) {
return false;
}
// Make sure the start bit rate is sane...
if (send_codec->startBitrate > 1000000) {
return false;
}
if (send_codec->codecType == kVideoCodecUnknown) {
return false;
}
number_of_cores_ = number_of_cores;
max_payload_size_ = max_payload_size;
memcpy(&send_codec_, send_codec, sizeof(VideoCodec));
if (send_codec_.maxBitrate == 0) {
// max is one bit per pixel
send_codec_.maxBitrate = (static_cast<int>(send_codec_.height) *
static_cast<int>(send_codec_.width) *
static_cast<int>(send_codec_.maxFramerate)) / 1000;
if (send_codec_.startBitrate > send_codec_.maxBitrate) {
// But if the user tries to set a higher start bit rate we will
// increase the max accordingly.
send_codec_.maxBitrate = send_codec_.startBitrate;
}
}
return true;
}
bool VCMCodecDataBase::SendCodec(VideoCodec* current_send_codec) const {
WEBRTC_TRACE(webrtc::kTraceApiCall, webrtc::kTraceVideoCoding, VCMId(id_),
"SendCodec");
if (!ptr_encoder_) {
return false;
}
memcpy(current_send_codec, &send_codec_, sizeof(VideoCodec));
return true;
}
VideoCodecType VCMCodecDataBase::SendCodec() const {
WEBRTC_TRACE(webrtc::kTraceApiCall, webrtc::kTraceVideoCoding, VCMId(id_),
"SendCodec type");
if (!ptr_encoder_) {
return kVideoCodecUnknown;
}
return send_codec_.codecType;
}
bool VCMCodecDataBase::DeregisterExternalEncoder(
uint8_t payload_type, bool* was_send_codec) {
assert(was_send_codec);
*was_send_codec = false;
if (external_payload_type_ != payload_type) {
return false;
}
if (send_codec_.plType == payload_type) {
// De-register as send codec if needed.
DeleteEncoder();
memset(&send_codec_, 0, sizeof(VideoCodec));
current_enc_is_external_ = false;
*was_send_codec = true;
}
external_payload_type_ = 0;
external_encoder_ = NULL;
internal_source_ = false;
return true;
}
void VCMCodecDataBase::RegisterExternalEncoder(
VideoEncoder* external_encoder,
uint8_t payload_type,
bool internal_source) {
// Since only one encoder can be used at a given time, only one external
// encoder can be registered/used.
external_encoder_ = external_encoder;
external_payload_type_ = payload_type;
internal_source_ = internal_source;
}
VCMGenericEncoder* VCMCodecDataBase::GetEncoder(
const VideoCodec* settings,
VCMEncodedFrameCallback* encoded_frame_callback) {
// If encoder exists, will destroy it and create new one.
DeleteEncoder();
if (settings->plType == external_payload_type_) {
// External encoder.
ptr_encoder_ = new VCMGenericEncoder(*external_encoder_, internal_source_);
current_enc_is_external_ = true;
} else {
ptr_encoder_ = CreateEncoder(settings->codecType);
current_enc_is_external_ = false;
}
encoded_frame_callback->SetPayloadType(settings->plType);
if (!ptr_encoder_) {
WEBRTC_TRACE(webrtc::kTraceError,
webrtc::kTraceVideoCoding,
VCMId(id_),
"Failed to create encoder: %s.",
settings->plName);
return NULL;
}
if (ptr_encoder_->InitEncode(settings, number_of_cores_, max_payload_size_) <
0) {
WEBRTC_TRACE(webrtc::kTraceError,
webrtc::kTraceVideoCoding,
VCMId(id_),
"Failed to initialize encoder: %s.",
settings->plName);
DeleteEncoder();
return NULL;
} else if (ptr_encoder_->RegisterEncodeCallback(encoded_frame_callback) <
0) {
DeleteEncoder();
return NULL;
}
// Intentionally don't check return value since the encoder registration
// shouldn't fail because the codec doesn't support changing the periodic key
// frame setting.
ptr_encoder_->SetPeriodicKeyFrames(periodic_key_frames_);
return ptr_encoder_;
}
bool VCMCodecDataBase::SetPeriodicKeyFrames(bool enable) {
periodic_key_frames_ = enable;
if (ptr_encoder_) {
return (ptr_encoder_->SetPeriodicKeyFrames(periodic_key_frames_) == 0);
}
return true;
}
void VCMCodecDataBase::ResetReceiver() {
ReleaseDecoder(ptr_decoder_);
ptr_decoder_ = NULL;
memset(&receive_codec_, 0, sizeof(VideoCodec));
while (!dec_map_.empty()) {
DecoderMap::iterator it = dec_map_.begin();
delete (*it).second;
dec_map_.erase(it);
}
while (!dec_external_map_.empty()) {
ExternalDecoderMap::iterator external_it = dec_external_map_.begin();
delete (*external_it).second;
dec_external_map_.erase(external_it);
}
current_dec_is_external_ = false;
}
bool VCMCodecDataBase::DeregisterExternalDecoder(uint8_t payload_type) {
ExternalDecoderMap::iterator it = dec_external_map_.find(payload_type);
if (it == dec_external_map_.end()) {
// Not found
return false;
}
if (receive_codec_.plType == payload_type) {
// Release it if it was registered and in use.
ReleaseDecoder(ptr_decoder_);
ptr_decoder_ = NULL;
}
DeregisterReceiveCodec(payload_type);
delete (*it).second;
dec_external_map_.erase(it);
return true;
}
// Add the external encoder object to the list of external decoders.
// Won't be registered as a receive codec until RegisterReceiveCodec is called.
bool VCMCodecDataBase::RegisterExternalDecoder(
VideoDecoder* external_decoder,
uint8_t payload_type,
bool internal_render_timing) {
// Check if payload value already exists, if so - erase old and insert new.
VCMExtDecoderMapItem* ext_decoder = new VCMExtDecoderMapItem(
external_decoder, payload_type, internal_render_timing);
if (!ext_decoder) {
return false;
}
DeregisterExternalDecoder(payload_type);
dec_external_map_[payload_type] = ext_decoder;
return true;
}
bool VCMCodecDataBase::DecoderRegistered() const {
return !dec_map_.empty();
}
bool VCMCodecDataBase::RegisterReceiveCodec(
const VideoCodec* receive_codec,
int number_of_cores,
bool require_key_frame) {
if (number_of_cores < 0) {
return false;
}
WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCoding, VCMId(id_),
"Codec: %s, Payload type %d, Height %d, Width %d, Bitrate %d,"
"Framerate %d.",
receive_codec->plName, receive_codec->plType,
receive_codec->height, receive_codec->width,
receive_codec->startBitrate, receive_codec->maxFramerate);
// Check if payload value already exists, if so - erase old and insert new.
DeregisterReceiveCodec(receive_codec->plType);
if (receive_codec->codecType == kVideoCodecUnknown) {
return false;
}
VideoCodec* new_receive_codec = new VideoCodec(*receive_codec);
dec_map_[receive_codec->plType] = new VCMDecoderMapItem(new_receive_codec,
number_of_cores,
require_key_frame);
return true;
}
bool VCMCodecDataBase::DeregisterReceiveCodec(
uint8_t payload_type) {
DecoderMap::iterator it = dec_map_.find(payload_type);
if (it == dec_map_.end()) {
return false;
}
VCMDecoderMapItem* dec_item = (*it).second;
delete dec_item;
dec_map_.erase(it);
if (receive_codec_.plType == payload_type) {
// This codec is currently in use.
memset(&receive_codec_, 0, sizeof(VideoCodec));
current_dec_is_external_ = false;
}
return true;
}
bool VCMCodecDataBase::ReceiveCodec(VideoCodec* current_receive_codec) const {
assert(current_receive_codec);
if (!ptr_decoder_) {
return false;
}
memcpy(current_receive_codec, &receive_codec_, sizeof(VideoCodec));
return true;
}
VideoCodecType VCMCodecDataBase::ReceiveCodec() const {
if (!ptr_decoder_) {
return kVideoCodecUnknown;
}
return receive_codec_.codecType;
}
VCMGenericDecoder* VCMCodecDataBase::GetDecoder(
uint8_t payload_type, VCMDecodedFrameCallback* decoded_frame_callback) {
if (payload_type == receive_codec_.plType || payload_type == 0) {
return ptr_decoder_;
}
// Check for exisitng decoder, if exists - delete.
if (ptr_decoder_) {
ReleaseDecoder(ptr_decoder_);
ptr_decoder_ = NULL;
memset(&receive_codec_, 0, sizeof(VideoCodec));
}
ptr_decoder_ = CreateAndInitDecoder(payload_type, &receive_codec_,
&current_dec_is_external_);
if (!ptr_decoder_) {
return NULL;
}
if (ptr_decoder_->RegisterDecodeCompleteCallback(decoded_frame_callback)
< 0) {
ReleaseDecoder(ptr_decoder_);
ptr_decoder_ = NULL;
memset(&receive_codec_, 0, sizeof(VideoCodec));
return NULL;
}
return ptr_decoder_;
}
VCMGenericDecoder* VCMCodecDataBase::CreateDecoderCopy() const {
if (!ptr_decoder_) {
return NULL;
}
VideoDecoder* decoder_copy = ptr_decoder_->_decoder.Copy();
if (!decoder_copy) {
return NULL;
}
return new VCMGenericDecoder(*decoder_copy, id_, ptr_decoder_->External());
}
void VCMCodecDataBase::ReleaseDecoder(VCMGenericDecoder* decoder) const {
if (decoder) {
assert(&decoder->_decoder);
decoder->Release();
if (!decoder->External()) {
delete &decoder->_decoder;
}
delete decoder;
}
}
void VCMCodecDataBase::CopyDecoder(const VCMGenericDecoder& decoder) {
VideoDecoder* decoder_copy = decoder._decoder.Copy();
if (decoder_copy) {
VCMDecodedFrameCallback* cb = ptr_decoder_->_callback;
ReleaseDecoder(ptr_decoder_);
ptr_decoder_ = new VCMGenericDecoder(*decoder_copy, id_,
decoder.External());
if (cb && ptr_decoder_->RegisterDecodeCompleteCallback(cb)) {
assert(false);
}
}
}
bool VCMCodecDataBase::SupportsRenderScheduling() const {
bool render_timing = true;
if (current_dec_is_external_) {
const VCMExtDecoderMapItem* ext_item = FindExternalDecoderItem(
receive_codec_.plType);
render_timing = ext_item->internal_render_timing;
}
return render_timing;
}
VCMGenericDecoder* VCMCodecDataBase::CreateAndInitDecoder(
uint8_t payload_type,
VideoCodec* new_codec,
bool* external) const {
assert(external);
assert(new_codec);
const VCMDecoderMapItem* decoder_item = FindDecoderItem(payload_type);
if (!decoder_item) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding, VCMId(id_),
"Unknown payload type: %u", payload_type);
return NULL;
}
VCMGenericDecoder* ptr_decoder = NULL;
const VCMExtDecoderMapItem* external_dec_item = FindExternalDecoderItem(
payload_type);
if (external_dec_item) {
// External codec.
ptr_decoder = new VCMGenericDecoder(
*external_dec_item->external_decoder_instance, id_, true);
*external = true;
} else {
// Create decoder.
ptr_decoder = CreateDecoder(decoder_item->settings->codecType);
*external = false;
}
if (!ptr_decoder) {
return NULL;
}
if (ptr_decoder->InitDecode(decoder_item->settings.get(),
decoder_item->number_of_cores,
decoder_item->require_key_frame) < 0) {
ReleaseDecoder(ptr_decoder);
return NULL;
}
memcpy(new_codec, decoder_item->settings.get(), sizeof(VideoCodec));
return ptr_decoder;
}
VCMGenericEncoder* VCMCodecDataBase::CreateEncoder(
const VideoCodecType type) const {
switch (type) {
#ifdef VIDEOCODEC_VP8
case kVideoCodecVP8:
return new VCMGenericEncoder(*(VP8Encoder::Create()));
#endif
#ifdef VIDEOCODEC_I420
case kVideoCodecI420:
return new VCMGenericEncoder(*(new I420Encoder));
#endif
default:
return NULL;
}
}
void VCMCodecDataBase::DeleteEncoder() {
if (ptr_encoder_) {
ptr_encoder_->Release();
if (!current_enc_is_external_) {
delete &ptr_encoder_->_encoder;
}
delete ptr_encoder_;
ptr_encoder_ = NULL;
}
}
VCMGenericDecoder* VCMCodecDataBase::CreateDecoder(VideoCodecType type) const {
switch (type) {
#ifdef VIDEOCODEC_VP8
case kVideoCodecVP8:
return new VCMGenericDecoder(*(VP8Decoder::Create()), id_);
#endif
#ifdef VIDEOCODEC_I420
case kVideoCodecI420:
return new VCMGenericDecoder(*(new I420Decoder), id_);
#endif
default:
return NULL;
}
}
const VCMDecoderMapItem* VCMCodecDataBase::FindDecoderItem(
uint8_t payload_type) const {
DecoderMap::const_iterator it = dec_map_.find(payload_type);
if (it != dec_map_.end()) {
return (*it).second;
}
return NULL;
}
const VCMExtDecoderMapItem* VCMCodecDataBase::FindExternalDecoderItem(
uint8_t payload_type) const {
ExternalDecoderMap::const_iterator it = dec_external_map_.find(payload_type);
if (it != dec_external_map_.end()) {
return (*it).second;
}
return NULL;
}
} // namespace webrtc

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_CODEC_DATABASE_H_
#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_CODEC_DATABASE_H_
#include <map>
#include "modules/video_coding/codecs/interface/video_codec_interface.h"
#include "modules/video_coding/main/interface/video_coding.h"
#include "modules/video_coding/main/source/generic_decoder.h"
#include "modules/video_coding/main/source/generic_encoder.h"
#include "system_wrappers/interface/scoped_ptr.h"
#include "typedefs.h"
namespace webrtc {
enum VCMCodecDBProperties {
kDefaultPayloadSize = 1440
};
struct VCMDecoderMapItem {
public:
VCMDecoderMapItem(VideoCodec* settings,
int number_of_cores,
bool require_key_frame);
scoped_ptr<VideoCodec> settings;
int number_of_cores;
bool require_key_frame;
};
struct VCMExtDecoderMapItem {
public:
VCMExtDecoderMapItem(VideoDecoder* external_decoder_instance,
uint8_t payload_type,
bool internal_render_timing);
uint8_t payload_type;
VideoDecoder* external_decoder_instance;
bool internal_render_timing;
};
class VCMCodecDataBase {
public:
explicit VCMCodecDataBase(int id);
~VCMCodecDataBase();
// Sender Side
// Returns the number of supported codecs (or -1 in case of error).
static int NumberOfCodecs();
// Returns the default settings for the codec with id |list_id|.
static bool Codec(int list_id, VideoCodec* settings);
// Returns the default settings for the codec with type |codec_type|.
static bool Codec(VideoCodecType codec_type, VideoCodec* settings);
void ResetSender();
// Sets the sender side codec and initiates the desired codec given the
// VideoCodec struct.
// Returns true if the codec was successfully registered, false otherwise.
bool RegisterSendCodec(const VideoCodec* send_codec,
int number_of_cores,
int max_payload_size);
// Gets the current send codec. Relevant for internal codecs only.
// Returns true if there is a send codec, false otherwise.
bool SendCodec(VideoCodec* current_send_codec) const;
// Gets current send side codec type. Relevant for internal codecs only.
// Returns kVideoCodecUnknown if there is no send codec.
VideoCodecType SendCodec() const;
// Registers and initializes an external encoder object.
// |internal_source| should be set to true if the codec has an internal
// video source and doesn't need the user to provide it with frames via
// the Encode() method.
void RegisterExternalEncoder(VideoEncoder* external_encoder,
uint8_t payload_type,
bool internal_source);
// Deregisters an external encoder. Returns true if the encoder was
// found and deregistered, false otherwise. |was_send_codec| is set to true
// if the external encoder was the send codec before being deregistered.
bool DeregisterExternalEncoder(uint8_t payload_type, bool* was_send_codec);
// Returns an encoder specified by the payload type in |settings|. The
// encoded frame callback of the encoder is set to |encoded_frame_callback|.
// If no such encoder already exists an instance will be created and
// initialized using |settings|.
// NULL is returned if no encoder with the specified payload type was found
// and the function failed to create one.
VCMGenericEncoder* GetEncoder(
const VideoCodec* settings,
VCMEncodedFrameCallback* encoded_frame_callback);
bool SetPeriodicKeyFrames(bool enable);
// Receiver Side
void ResetReceiver();
// Deregisters an external decoder object specified by |payload_type|.
bool DeregisterExternalDecoder(uint8_t payload_type);
// Registers an external decoder object to the payload type |payload_type|.
// |internal_render_timing| is set to true if the |external_decoder| has
// built in rendering which is able to obey the render timestamps of the
// encoded frames.
bool RegisterExternalDecoder(VideoDecoder* external_decoder,
uint8_t payload_type,
bool internal_render_timing);
bool DecoderRegistered() const;
bool RegisterReceiveCodec(const VideoCodec* receive_codec,
int number_of_cores,
bool require_key_frame);
bool DeregisterReceiveCodec(uint8_t payload_type);
// Get current receive side codec. Relevant for internal codecs only.
bool ReceiveCodec(VideoCodec* current_receive_codec) const;
// Get current receive side codec type. Relevant for internal codecs only.
VideoCodecType ReceiveCodec() const;
// Returns a decoder specified by |payload_type|. The decoded frame callback
// of the encoder is set to |decoded_frame_callback|. If no such decoder
// already exists an instance will be created and initialized.
// NULL is returned if no encoder with the specified payload type was found
// and the function failed to create one.
VCMGenericDecoder* GetDecoder(
uint8_t payload_type, VCMDecodedFrameCallback* decoded_frame_callback);
// Returns a deep copy of the currently active decoder.
VCMGenericDecoder* CreateDecoderCopy() const;
// Deletes the memory of the decoder instance |decoder|. Used to delete
// deep copies returned by CreateDecoderCopy().
void ReleaseDecoder(VCMGenericDecoder* decoder) const;
// Creates a deep copy of |decoder| and replaces the currently used decoder
// with it.
void CopyDecoder(const VCMGenericDecoder& decoder);
// Returns true if the currently active decoder supports render scheduling,
// that is, it is able to render frames according to the render timestamp of
// the encoded frames.
bool SupportsRenderScheduling() const;
private:
typedef std::map<uint8_t, VCMDecoderMapItem*> DecoderMap;
typedef std::map<uint8_t, VCMExtDecoderMapItem*> ExternalDecoderMap;
VCMGenericDecoder* CreateAndInitDecoder(uint8_t payload_type,
VideoCodec* new_codec,
bool* external) const;
// Create an internal encoder given a codec type.
VCMGenericEncoder* CreateEncoder(const VideoCodecType type) const;
void DeleteEncoder();
// Create an internal Decoder given a codec type
VCMGenericDecoder* CreateDecoder(VideoCodecType type) const;
const VCMDecoderMapItem* FindDecoderItem(uint8_t payload_type) const;
const VCMExtDecoderMapItem* FindExternalDecoderItem(
uint8_t payload_type) const;
int id_;
int number_of_cores_;
int max_payload_size_;
bool periodic_key_frames_;
bool current_enc_is_external_;
VideoCodec send_codec_;
VideoCodec receive_codec_;
uint8_t external_payload_type_;
VideoEncoder* external_encoder_;
bool internal_source_;
VCMGenericEncoder* ptr_encoder_;
VCMGenericDecoder* ptr_decoder_;
bool current_dec_is_external_;
DecoderMap dec_map_;
ExternalDecoderMap dec_external_map_;
}; // VCMCodecDataBase
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_CODEC_DATABASE_H_

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "codec_timer.h"
#include <assert.h>
namespace webrtc
{
VCMCodecTimer::VCMCodecTimer()
:
_filteredMax(0),
_firstDecodeTime(true),
_shortMax(0),
_history()
{
Reset();
}
WebRtc_Word32 VCMCodecTimer::StopTimer(WebRtc_Word64 startTimeMs, WebRtc_Word64 nowMs)
{
const WebRtc_Word32 timeDiff = static_cast<WebRtc_Word32>(nowMs - startTimeMs);
MaxFilter(timeDiff, nowMs);
return timeDiff;
}
void VCMCodecTimer::Reset()
{
_filteredMax = 0;
_firstDecodeTime = true;
_shortMax = 0;
for (int i=0; i < MAX_HISTORY_SIZE; i++)
{
_history[i].shortMax = 0;
_history[i].timeMs = -1;
}
}
// Update the max-value filter
void VCMCodecTimer::MaxFilter(WebRtc_Word32 decodeTime, WebRtc_Word64 nowMs)
{
if (!_firstDecodeTime)
{
UpdateMaxHistory(decodeTime, nowMs);
ProcessHistory(nowMs);
}
else
{
_firstDecodeTime = false;
}
}
void
VCMCodecTimer::UpdateMaxHistory(WebRtc_Word32 decodeTime, WebRtc_Word64 now)
{
if (_history[0].timeMs >= 0 &&
now - _history[0].timeMs < SHORT_FILTER_MS)
{
if (decodeTime > _shortMax)
{
_shortMax = decodeTime;
}
}
else
{
// Only add a new value to the history once a second
if(_history[0].timeMs == -1)
{
// First, no shift
_shortMax = decodeTime;
}
else
{
// Shift
for(int i = (MAX_HISTORY_SIZE - 2); i >= 0 ; i--)
{
_history[i+1].shortMax = _history[i].shortMax;
_history[i+1].timeMs = _history[i].timeMs;
}
}
if (_shortMax == 0)
{
_shortMax = decodeTime;
}
_history[0].shortMax = _shortMax;
_history[0].timeMs = now;
_shortMax = 0;
}
}
void
VCMCodecTimer::ProcessHistory(WebRtc_Word64 nowMs)
{
_filteredMax = _shortMax;
if (_history[0].timeMs == -1)
{
return;
}
for (int i=0; i < MAX_HISTORY_SIZE; i++)
{
if (_history[i].timeMs == -1)
{
break;
}
if (nowMs - _history[i].timeMs > MAX_HISTORY_SIZE * SHORT_FILTER_MS)
{
// This sample (and all samples after this) is too old
break;
}
if (_history[i].shortMax > _filteredMax)
{
// This sample is the largest one this far into the history
_filteredMax = _history[i].shortMax;
}
}
}
// Get the maximum observed time within a time window
WebRtc_Word32 VCMCodecTimer::RequiredDecodeTimeMs(FrameType /*frameType*/) const
{
return _filteredMax;
}
}

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
#include "typedefs.h"
#include "module_common_types.h"
namespace webrtc
{
// MAX_HISTORY_SIZE * SHORT_FILTER_MS defines the window size in milliseconds
#define MAX_HISTORY_SIZE 20
#define SHORT_FILTER_MS 1000
class VCMShortMaxSample
{
public:
VCMShortMaxSample() : shortMax(0), timeMs(-1) {};
WebRtc_Word32 shortMax;
WebRtc_Word64 timeMs;
};
class VCMCodecTimer
{
public:
VCMCodecTimer();
// Updates and returns the max filtered decode time.
WebRtc_Word32 StopTimer(WebRtc_Word64 startTimeMs, WebRtc_Word64 nowMs);
// Empty the list of timers.
void Reset();
// Get the required decode time in ms.
WebRtc_Word32 RequiredDecodeTimeMs(FrameType frameType) const;
private:
void UpdateMaxHistory(WebRtc_Word32 decodeTime, WebRtc_Word64 now);
void MaxFilter(WebRtc_Word32 newTime, WebRtc_Word64 nowMs);
void ProcessHistory(WebRtc_Word64 nowMs);
WebRtc_Word32 _filteredMax;
bool _firstDecodeTime;
WebRtc_Word32 _shortMax;
VCMShortMaxSample _history[MAX_HISTORY_SIZE];
};
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/video_coding/main/source/content_metrics_processing.h"
#include <math.h>
#include "modules/interface/module_common_types.h"
#include "modules/video_coding/main/interface/video_coding_defines.h"
namespace webrtc {
//////////////////////////////////
/// VCMContentMetricsProcessing //
//////////////////////////////////
VCMContentMetricsProcessing::VCMContentMetricsProcessing()
: recursive_avg_factor_(1 / 150.0f), // matched to 30fps.
frame_cnt_uniform_avg_(0),
avg_motion_level_(0.0f),
avg_spatial_level_(0.0f) {
recursive_avg_ = new VideoContentMetrics();
uniform_avg_ = new VideoContentMetrics();
}
VCMContentMetricsProcessing::~VCMContentMetricsProcessing() {
delete recursive_avg_;
delete uniform_avg_;
}
int VCMContentMetricsProcessing::Reset() {
recursive_avg_->Reset();
uniform_avg_->Reset();
frame_cnt_uniform_avg_ = 0;
avg_motion_level_ = 0.0f;
avg_spatial_level_ = 0.0f;
return VCM_OK;
}
void VCMContentMetricsProcessing::UpdateFrameRate(uint32_t frameRate) {
// Update factor for recursive averaging.
recursive_avg_factor_ = static_cast<float> (1000.0f) /
static_cast<float>(frameRate * kQmMinIntervalMs);
}
VideoContentMetrics* VCMContentMetricsProcessing::LongTermAvgData() {
return recursive_avg_;
}
VideoContentMetrics* VCMContentMetricsProcessing::ShortTermAvgData() {
if (frame_cnt_uniform_avg_ == 0) {
return NULL;
}
// Two metrics are used: motion and spatial level.
uniform_avg_->motion_magnitude = avg_motion_level_ /
static_cast<float>(frame_cnt_uniform_avg_);
uniform_avg_->spatial_pred_err = avg_spatial_level_ /
static_cast<float>(frame_cnt_uniform_avg_);
return uniform_avg_;
}
void VCMContentMetricsProcessing::ResetShortTermAvgData() {
// Reset.
avg_motion_level_ = 0.0f;
avg_spatial_level_ = 0.0f;
frame_cnt_uniform_avg_ = 0;
}
int VCMContentMetricsProcessing::UpdateContentData(
const VideoContentMetrics *contentMetrics) {
if (contentMetrics == NULL) {
return VCM_OK;
}
return ProcessContent(contentMetrics);
}
int VCMContentMetricsProcessing::ProcessContent(
const VideoContentMetrics *contentMetrics) {
// Update the recursive averaged metrics: average is over longer window
// of time: over QmMinIntervalMs ms.
UpdateRecursiveAvg(contentMetrics);
// Update the uniform averaged metrics: average is over shorter window
// of time: based on ~RTCP reports.
UpdateUniformAvg(contentMetrics);
return VCM_OK;
}
void VCMContentMetricsProcessing::UpdateUniformAvg(
const VideoContentMetrics *contentMetrics) {
// Update frame counter.
frame_cnt_uniform_avg_ += 1;
// Update averaged metrics: motion and spatial level are used.
avg_motion_level_ += contentMetrics->motion_magnitude;
avg_spatial_level_ += contentMetrics->spatial_pred_err;
return;
}
void VCMContentMetricsProcessing::UpdateRecursiveAvg(
const VideoContentMetrics *contentMetrics) {
// Spatial metrics: 2x2, 1x2(H), 2x1(V).
recursive_avg_->spatial_pred_err = (1 - recursive_avg_factor_) *
recursive_avg_->spatial_pred_err +
recursive_avg_factor_ * contentMetrics->spatial_pred_err;
recursive_avg_->spatial_pred_err_h = (1 - recursive_avg_factor_) *
recursive_avg_->spatial_pred_err_h +
recursive_avg_factor_ * contentMetrics->spatial_pred_err_h;
recursive_avg_->spatial_pred_err_v = (1 - recursive_avg_factor_) *
recursive_avg_->spatial_pred_err_v +
recursive_avg_factor_ * contentMetrics->spatial_pred_err_v;
// Motion metric: Derived from NFD (normalized frame difference).
recursive_avg_->motion_magnitude = (1 - recursive_avg_factor_) *
recursive_avg_->motion_magnitude +
recursive_avg_factor_ * contentMetrics->motion_magnitude;
}
} // end of namespace

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_CONTENT_METRICS_PROCESSING_H_
#define WEBRTC_MODULES_VIDEO_CODING_CONTENT_METRICS_PROCESSING_H_
#include "typedefs.h"
namespace webrtc {
struct VideoContentMetrics;
// QM interval time (in ms)
enum {
kQmMinIntervalMs = 10000
};
// Flag for NFD metric vs motion metric
enum {
kNfdMetric = 1
};
/**********************************/
/* Content Metrics Processing */
/**********************************/
class VCMContentMetricsProcessing {
public:
VCMContentMetricsProcessing();
~VCMContentMetricsProcessing();
// Update class with latest metrics.
int UpdateContentData(const VideoContentMetrics *contentMetrics);
// Reset the short-term averaged content data.
void ResetShortTermAvgData();
// Initialize.
int Reset();
// Inform class of current frame rate.
void UpdateFrameRate(uint32_t frameRate);
// Returns the long-term averaged content data: recursive average over longer
// time scale.
VideoContentMetrics* LongTermAvgData();
// Returns the short-term averaged content data: uniform average over
// shorter time scalE.
VideoContentMetrics* ShortTermAvgData();
private:
// Compute working average.
int ProcessContent(const VideoContentMetrics *contentMetrics);
// Update the recursive averaged metrics: longer time average (~5/10 secs).
void UpdateRecursiveAvg(const VideoContentMetrics *contentMetrics);
// Update the uniform averaged metrics: shorter time average (~RTCP report).
void UpdateUniformAvg(const VideoContentMetrics *contentMetrics);
VideoContentMetrics* recursive_avg_;
VideoContentMetrics* uniform_avg_;
float recursive_avg_factor_;
uint32_t frame_cnt_uniform_avg_;
float avg_motion_level_;
float avg_spatial_level_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_CONTENT_METRICS_PROCESSING_H_

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/video_coding/main/source/decoding_state.h"
#include "modules/video_coding/main/source/frame_buffer.h"
#include "modules/video_coding/main/source/jitter_buffer_common.h"
#include "modules/video_coding/main/source/packet.h"
#include "modules/interface/module_common_types.h"
namespace webrtc {
VCMDecodingState::VCMDecodingState()
: sequence_num_(0),
time_stamp_(0),
picture_id_(kNoPictureId),
temporal_id_(kNoTemporalIdx),
tl0_pic_id_(kNoTl0PicIdx),
full_sync_(true),
init_(true) {}
VCMDecodingState::~VCMDecodingState() {}
void VCMDecodingState::Reset() {
// TODO(mikhal): Verify - not always would want to reset the sync
sequence_num_ = 0;
time_stamp_ = 0;
picture_id_ = kNoPictureId;
temporal_id_ = kNoTemporalIdx;
tl0_pic_id_ = kNoTl0PicIdx;
full_sync_ = true;
init_ = true;
}
uint32_t VCMDecodingState::time_stamp() const {
return time_stamp_;
}
uint16_t VCMDecodingState::sequence_num() const {
return sequence_num_;
}
bool VCMDecodingState::IsOldFrame(const VCMFrameBuffer* frame) const {
assert(frame != NULL);
if (init_)
return false;
return (LatestTimestamp(time_stamp_, frame->TimeStamp(), NULL)
== time_stamp_);
}
bool VCMDecodingState::IsOldPacket(const VCMPacket* packet) const {
assert(packet != NULL);
if (init_)
return false;
return (LatestTimestamp(time_stamp_, packet->timestamp, NULL)
== time_stamp_);
}
void VCMDecodingState::SetState(const VCMFrameBuffer* frame) {
assert(frame != NULL && frame->GetHighSeqNum() >= 0);
UpdateSyncState(frame);
sequence_num_ = static_cast<uint16_t>(frame->GetHighSeqNum());
time_stamp_ = frame->TimeStamp();
picture_id_ = frame->PictureId();
temporal_id_ = frame->TemporalId();
tl0_pic_id_ = frame->Tl0PicId();
init_ = false;
}
void VCMDecodingState::SetStateOneBack(const VCMFrameBuffer* frame) {
assert(frame != NULL && frame->GetHighSeqNum() >= 0);
sequence_num_ = static_cast<uint16_t>(frame->GetHighSeqNum()) - 1u;
time_stamp_ = frame->TimeStamp() - 1u;
temporal_id_ = frame->TemporalId();
if (frame->PictureId() != kNoPictureId) {
if (frame->PictureId() == 0)
picture_id_ = 0x7FFF;
else
picture_id_ = frame->PictureId() - 1;
}
if (frame->Tl0PicId() != kNoTl0PicIdx) {
if (frame->Tl0PicId() == 0)
tl0_pic_id_ = 0x00FF;
else
tl0_pic_id_ = frame->Tl0PicId() - 1;
}
init_ = false;
}
void VCMDecodingState::UpdateOldPacket(const VCMPacket* packet) {
assert(packet != NULL);
if (packet->timestamp == time_stamp_) {
// Late packet belonging to the last decoded frame - make sure we update the
// last decoded sequence number.
sequence_num_ = LatestSequenceNumber(packet->seqNum, sequence_num_, NULL);
}
}
void VCMDecodingState::SetSeqNum(uint16_t new_seq_num) {
sequence_num_ = new_seq_num;
}
bool VCMDecodingState::init() const {
return init_;
}
bool VCMDecodingState::full_sync() const {
return full_sync_;
}
void VCMDecodingState::UpdateSyncState(const VCMFrameBuffer* frame) {
if (init_)
return;
if (frame->TemporalId() == kNoTemporalIdx ||
frame->Tl0PicId() == kNoTl0PicIdx) {
full_sync_ = true;
} else if (frame->FrameType() == kVideoFrameKey || frame->LayerSync()) {
full_sync_ = true;
} else if (full_sync_) {
// Verify that we are still in sync.
// Sync will be broken if continuity is true for layers but not for the
// other methods (PictureId and SeqNum).
if (UsingPictureId(frame)) {
full_sync_ = ContinuousPictureId(frame->PictureId());
} else {
full_sync_ = ContinuousSeqNum(static_cast<uint16_t>(
frame->GetLowSeqNum()));
}
}
}
bool VCMDecodingState::ContinuousFrame(const VCMFrameBuffer* frame) const {
// Check continuity based on the following hierarchy:
// - Temporal layers (stop here if out of sync).
// - Picture Id when available.
// - Sequence numbers.
// Return true when in initial state.
// Note that when a method is not applicable it will return false.
assert(frame != NULL);
if (init_)
return true;
if (!ContinuousLayer(frame->TemporalId(), frame->Tl0PicId())) {
// Base layers are not continuous or temporal layers are inactive.
// In the presence of temporal layers, check for Picture ID/sequence number
// continuity if sync can be restored by this frame.
if (!full_sync_ && !frame->LayerSync())
return false;
else if (UsingPictureId(frame)) {
return ContinuousPictureId(frame->PictureId());
} else {
return ContinuousSeqNum(static_cast<uint16_t>(frame->GetLowSeqNum()));
}
}
return true;
}
bool VCMDecodingState::ContinuousPictureId(int picture_id) const {
int next_picture_id = picture_id_ + 1;
if (picture_id < picture_id_) {
// Wrap
if (picture_id_ >= 0x80) {
// 15 bits used for picture id
return ((next_picture_id & 0x7FFF) == picture_id);
} else {
// 7 bits used for picture id
return ((next_picture_id & 0x7F) == picture_id);
}
}
// No wrap
return (next_picture_id == picture_id);
}
bool VCMDecodingState::ContinuousSeqNum(uint16_t seq_num) const {
return (seq_num == static_cast<uint16_t>(sequence_num_ + 1));
}
bool VCMDecodingState::ContinuousLayer(int temporal_id,
int tl0_pic_id) const {
// First, check if applicable.
if (temporal_id == kNoTemporalIdx || tl0_pic_id == kNoTl0PicIdx)
return false;
// If this is the first frame to use temporal layers, make sure we start
// from base.
else if (tl0_pic_id_ == kNoTl0PicIdx && temporal_id_ == kNoTemporalIdx &&
temporal_id == 0)
return true;
// Current implementation: Look for base layer continuity.
if (temporal_id != 0)
return false;
return (static_cast<uint8_t>(tl0_pic_id_ + 1) == tl0_pic_id);
}
bool VCMDecodingState::UsingPictureId(const VCMFrameBuffer* frame) const {
return (frame->PictureId() != kNoPictureId && picture_id_ != kNoPictureId);
}
} // namespace webrtc

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_DECODING_STATE_H_
#define WEBRTC_MODULES_VIDEO_CODING_DECODING_STATE_H_
#include "typedefs.h"
namespace webrtc {
// Forward declarations
class VCMFrameBuffer;
class VCMPacket;
class VCMDecodingState {
public:
VCMDecodingState();
~VCMDecodingState();
// Check for old frame
bool IsOldFrame(const VCMFrameBuffer* frame) const;
// Check for old packet
bool IsOldPacket(const VCMPacket* packet) const;
// Check for frame continuity based on current decoded state. Use best method
// possible, i.e. temporal info, picture ID or sequence number.
bool ContinuousFrame(const VCMFrameBuffer* frame) const;
void SetState(const VCMFrameBuffer* frame);
// Set the decoding state one frame back.
void SetStateOneBack(const VCMFrameBuffer* frame);
// Update the sequence number if the timestamp matches current state and the
// sequence number is higher than the current one. This accounts for packets
// arriving late.
void UpdateOldPacket(const VCMPacket* packet);
void SetSeqNum(uint16_t new_seq_num);
void Reset();
uint32_t time_stamp() const;
uint16_t sequence_num() const;
// Return true if at initial state.
bool init() const;
// Return true when sync is on - decode all layers.
bool full_sync() const;
private:
void UpdateSyncState(const VCMFrameBuffer* frame);
// Designated continuity functions
bool ContinuousPictureId(int picture_id) const;
bool ContinuousSeqNum(uint16_t seq_num) const;
bool ContinuousLayer(int temporal_id, int tl0_pic_id) const;
bool UsingPictureId(const VCMFrameBuffer* frame) const;
// Keep state of last decoded frame.
// TODO(mikhal/stefan): create designated classes to handle these types.
uint16_t sequence_num_;
uint32_t time_stamp_;
int picture_id_;
int temporal_id_;
int tl0_pic_id_;
bool full_sync_; // Sync flag when temporal layers are used.
bool init_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_DECODING_STATE_H_

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <string.h>
#include "modules/video_coding/main/source/decoding_state.h"
#include "modules/video_coding/main/source/frame_buffer.h"
#include "gtest/gtest.h"
#include "modules/video_coding/main/source/jitter_buffer_common.h"
#include "modules/interface/module_common_types.h"
#include "modules/video_coding/main/source/packet.h"
namespace webrtc {
TEST(TestDecodingState, Sanity) {
VCMDecodingState dec_state;
dec_state.Reset();
EXPECT_TRUE(dec_state.init());
EXPECT_TRUE(dec_state.full_sync());
}
TEST(TestDecodingState, FrameContinuity) {
VCMDecodingState dec_state;
// Check that makes decision based on correct method.
VCMFrameBuffer frame;
frame.SetState(kStateEmpty);
VCMPacket* packet = new VCMPacket();
packet->isFirstPacket = 1;
packet->timestamp = 1;
packet->seqNum = 0xffff;
packet->frameType = kVideoFrameDelta;
packet->codecSpecificHeader.codec = kRTPVideoVP8;
packet->codecSpecificHeader.codecHeader.VP8.pictureId = 0x007F;
frame.InsertPacket(*packet, 0, false, 0);
// Should return true on init.
dec_state.Reset();
EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
dec_state.SetState(&frame);
frame.Reset();
// Use pictureId
packet->codecSpecificHeader.codecHeader.VP8.pictureId = 0x0002;
frame.InsertPacket(*packet, 0, false, 0);
EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
frame.Reset();
frame.SetState(kStateEmpty);
packet->codecSpecificHeader.codecHeader.VP8.pictureId = 0;
packet->seqNum = 10;
frame.InsertPacket(*packet, 0, false, 0);
EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
// Use sequence numbers.
packet->codecSpecificHeader.codecHeader.VP8.pictureId = kNoPictureId;
frame.Reset();
frame.SetState(kStateEmpty);
packet->seqNum = dec_state.sequence_num() - 1u;
frame.InsertPacket(*packet, 0, false, 0);
EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
frame.Reset();
frame.SetState(kStateEmpty);
packet->seqNum = dec_state.sequence_num() + 1u;
frame.InsertPacket(*packet, 0, false, 0);
// Insert another packet to this frame
packet->seqNum++;
frame.InsertPacket(*packet, 0, false, 0);
// Verify wrap.
EXPECT_EQ(dec_state.sequence_num(), 0xffff);
EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
dec_state.SetState(&frame);
// Insert packet with temporal info.
dec_state.Reset();
frame.Reset();
frame.SetState(kStateEmpty);
packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
packet->codecSpecificHeader.codecHeader.VP8.pictureId = 0;
packet->seqNum = 1;
packet->timestamp = 1;
EXPECT_TRUE(dec_state.full_sync());
frame.InsertPacket(*packet, 0, false, 0);
dec_state.SetState(&frame);
EXPECT_TRUE(dec_state.full_sync());
frame.Reset();
frame.SetState(kStateEmpty);
// 1 layer up - still good.
packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 1;
packet->codecSpecificHeader.codecHeader.VP8.pictureId = 1;
packet->seqNum = 2;
packet->timestamp = 2;
frame.InsertPacket(*packet, 0, false, 0);
EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
dec_state.SetState(&frame);
EXPECT_TRUE(dec_state.full_sync());
frame.Reset();
frame.SetState(kStateEmpty);
// Lost non-base layer packet => should update sync parameter.
packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 3;
packet->codecSpecificHeader.codecHeader.VP8.pictureId = 3;
packet->seqNum = 4;
packet->timestamp = 4;
frame.InsertPacket(*packet, 0, false, 0);
EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
// Now insert the next non-base layer (belonging to a next tl0PicId).
frame.Reset();
frame.SetState(kStateEmpty);
packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 1;
packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 2;
packet->codecSpecificHeader.codecHeader.VP8.pictureId = 4;
packet->seqNum = 5;
packet->timestamp = 5;
frame.InsertPacket(*packet, 0, false, 0);
// Checking continuity and not updating the state - this should not trigger
// an update of sync state.
EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
EXPECT_TRUE(dec_state.full_sync());
// Next base layer (dropped interim non-base layers) - should update sync.
frame.Reset();
frame.SetState(kStateEmpty);
packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 1;
packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
packet->codecSpecificHeader.codecHeader.VP8.pictureId = 5;
packet->seqNum = 6;
packet->timestamp = 6;
frame.InsertPacket(*packet, 0, false, 0);
EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
dec_state.SetState(&frame);
EXPECT_FALSE(dec_state.full_sync());
// Check wrap for temporal layers.
frame.Reset();
frame.SetState(kStateEmpty);
packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0x00FF;
packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
packet->codecSpecificHeader.codecHeader.VP8.pictureId = 6;
packet->seqNum = 7;
packet->timestamp = 7;
frame.InsertPacket(*packet, 0, false, 0);
dec_state.SetState(&frame);
EXPECT_FALSE(dec_state.full_sync());
frame.Reset();
frame.SetState(kStateEmpty);
packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0x0000;
packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
packet->codecSpecificHeader.codecHeader.VP8.pictureId = 7;
packet->seqNum = 8;
packet->timestamp = 8;
frame.InsertPacket(*packet, 0, false, 0);
EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
// The current frame is not continuous
dec_state.SetState(&frame);
EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
delete packet;
}
TEST(TestDecodingState, SetStateOneBack) {
VCMDecodingState dec_state;
VCMFrameBuffer frame;
frame.SetState(kStateEmpty);
VCMPacket* packet = new VCMPacket();
// Based on PictureId.
packet->frameType = kVideoFrameDelta;
packet->codecSpecificHeader.codec = kRTPVideoVP8;
packet->timestamp = 0;
packet->seqNum = 0;
packet->codecSpecificHeader.codecHeader.VP8.pictureId = 0;
packet->frameType = kVideoFrameDelta;
frame.InsertPacket(*packet, 0, false, 0);
dec_state.SetStateOneBack(&frame);
EXPECT_EQ(dec_state.sequence_num(), 0xFFFF);
// Check continuity.
EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
// Based on Temporal layers.
packet->timestamp = 0;
packet->seqNum = 0;
packet->codecSpecificHeader.codecHeader.VP8.pictureId = kNoPictureId;
packet->frameType = kVideoFrameDelta;
packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
frame.InsertPacket(*packet, 0, false, 0);
dec_state.SetStateOneBack(&frame);
// Check continuity
EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
delete packet;
}
TEST(TestDecodingState, UpdateOldPacket) {
VCMDecodingState dec_state;
// Update only if zero size and newer than previous.
// Should only update if the timeStamp match.
VCMFrameBuffer frame;
frame.SetState(kStateEmpty);
VCMPacket* packet = new VCMPacket();
packet->timestamp = 1;
packet->seqNum = 1;
packet->frameType = kVideoFrameDelta;
frame.InsertPacket(*packet, 0, false, 0);
dec_state.SetState(&frame);
EXPECT_EQ(dec_state.sequence_num(), 1);
// Insert an empty packet that does not belong to the same frame.
// => Sequence num should be the same.
packet->timestamp = 2;
dec_state.UpdateOldPacket(packet);
EXPECT_EQ(dec_state.sequence_num(), 1);
// Now insert empty packet belonging to the same frame.
packet->timestamp = 1;
packet->seqNum = 2;
packet->frameType = kFrameEmpty;
packet->sizeBytes = 0;
dec_state.UpdateOldPacket(packet);
EXPECT_EQ(dec_state.sequence_num(), 2);
// Now insert delta packet belonging to the same frame.
packet->timestamp = 1;
packet->seqNum = 3;
packet->frameType = kVideoFrameDelta;
packet->sizeBytes = 1400;
dec_state.UpdateOldPacket(packet);
EXPECT_EQ(dec_state.sequence_num(), 3);
// Insert a packet belonging to an older timestamp - should not update the
// sequence number.
packet->timestamp = 0;
packet->seqNum = 4;
packet->frameType = kFrameEmpty;
packet->sizeBytes = 0;
dec_state.UpdateOldPacket(packet);
EXPECT_EQ(dec_state.sequence_num(), 3);
delete packet;
}
TEST(TestDecodingState, MultiLayerBehavior) {
// Identify sync/non-sync when more than one layer.
VCMDecodingState dec_state;
// Identify packets belonging to old frames/packets.
// Set state for current frames.
// tl0PicIdx 0, temporal id 0.
VCMFrameBuffer frame;
VCMPacket* packet = new VCMPacket();
packet->frameType = kVideoFrameDelta;
packet->codecSpecificHeader.codec = kRTPVideoVP8;
frame.SetState(kStateEmpty);
packet->timestamp = 0;
packet->seqNum = 0;
packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
packet->codecSpecificHeader.codecHeader.VP8.pictureId = 0;
frame.InsertPacket(*packet, 0, false, 0);
dec_state.SetState(&frame);
// tl0PicIdx 0, temporal id 1.
frame.Reset();
frame.SetState(kStateEmpty);
packet->timestamp = 1;
packet->seqNum = 1;
packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 1;
packet->codecSpecificHeader.codecHeader.VP8.pictureId = 1;
frame.InsertPacket(*packet, 0, false, 0);
EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
dec_state.SetState(&frame);
EXPECT_TRUE(dec_state.full_sync());
// Lost tl0PicIdx 0, temporal id 2.
// Insert tl0PicIdx 0, temporal id 3.
frame.Reset();
frame.SetState(kStateEmpty);
packet->timestamp = 3;
packet->seqNum = 3;
packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 3;
packet->codecSpecificHeader.codecHeader.VP8.pictureId = 3;
frame.InsertPacket(*packet, 0, false, 0);
EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
dec_state.SetState(&frame);
EXPECT_FALSE(dec_state.full_sync());
// Insert next base layer
frame.Reset();
frame.SetState(kStateEmpty);
packet->timestamp = 4;
packet->seqNum = 4;
packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 1;
packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
packet->codecSpecificHeader.codecHeader.VP8.pictureId = 4;
frame.InsertPacket(*packet, 0, false, 0);
EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
dec_state.SetState(&frame);
EXPECT_FALSE(dec_state.full_sync());
// Insert key frame - should update sync value.
// A key frame is always a base layer.
frame.Reset();
frame.SetState(kStateEmpty);
packet->frameType = kVideoFrameKey;
packet->isFirstPacket = 1;
packet->timestamp = 5;
packet->seqNum = 5;
packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 2;
packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
packet->codecSpecificHeader.codecHeader.VP8.pictureId = 5;
frame.InsertPacket(*packet, 0, false, 0);
EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
dec_state.SetState(&frame);
EXPECT_TRUE(dec_state.full_sync());
// After sync, a continuous PictureId is required
// (continuous base layer is not enough )
frame.Reset();
frame.SetState(kStateEmpty);
packet->frameType = kVideoFrameDelta;
packet->timestamp = 6;
packet->seqNum = 6;
packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 3;
packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
packet->codecSpecificHeader.codecHeader.VP8.pictureId = 6;
frame.InsertPacket(*packet, 0, false, 0);
EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
EXPECT_TRUE(dec_state.full_sync());
frame.Reset();
frame.SetState(kStateEmpty);
packet->frameType = kVideoFrameDelta;
packet->isFirstPacket = 1;
packet->timestamp = 8;
packet->seqNum = 8;
packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 4;
packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
packet->codecSpecificHeader.codecHeader.VP8.pictureId = 8;
frame.InsertPacket(*packet, 0, false, 0);
EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
EXPECT_TRUE(dec_state.full_sync());
dec_state.SetState(&frame);
EXPECT_FALSE(dec_state.full_sync());
// Insert a non-ref frame - should update sync value.
frame.Reset();
frame.SetState(kStateEmpty);
packet->frameType = kVideoFrameDelta;
packet->isFirstPacket = 1;
packet->timestamp = 9;
packet->seqNum = 9;
packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 4;
packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 2;
packet->codecSpecificHeader.codecHeader.VP8.pictureId = 9;
packet->codecSpecificHeader.codecHeader.VP8.layerSync = true;
frame.InsertPacket(*packet, 0, false, 0);
dec_state.SetState(&frame);
EXPECT_TRUE(dec_state.full_sync());
// The following test will verify the sync flag behavior after a loss.
// Create the following pattern:
// Update base layer, lose packet 1 (sync flag on, layer 2), insert packet 3
// (sync flag on, layer 2) check continuity and sync flag after inserting
// packet 2 (sync flag on, layer 1).
// Base layer.
frame.Reset();
dec_state.Reset();
frame.SetState(kStateEmpty);
packet->frameType = kVideoFrameDelta;
packet->isFirstPacket = 1;
packet->markerBit = 1;
packet->timestamp = 0;
packet->seqNum = 0;
packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
packet->codecSpecificHeader.codecHeader.VP8.pictureId = 0;
packet->codecSpecificHeader.codecHeader.VP8.layerSync = false;
frame.InsertPacket(*packet, 0, false, 0);
dec_state.SetState(&frame);
EXPECT_TRUE(dec_state.full_sync());
// Layer 2 - 2 packets (insert one, lose one).
frame.Reset();
frame.SetState(kStateEmpty);
packet->frameType = kVideoFrameDelta;
packet->isFirstPacket = 1;
packet->markerBit = 0;
packet->timestamp = 1;
packet->seqNum = 1;
packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 2;
packet->codecSpecificHeader.codecHeader.VP8.pictureId = 1;
packet->codecSpecificHeader.codecHeader.VP8.layerSync = true;
frame.InsertPacket(*packet, 0, false, 0);
EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
// Layer 1
frame.Reset();
frame.SetState(kStateEmpty);
packet->frameType = kVideoFrameDelta;
packet->isFirstPacket = 1;
packet->markerBit = 1;
packet->timestamp = 2;
packet->seqNum = 3;
packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 1;
packet->codecSpecificHeader.codecHeader.VP8.pictureId = 2;
packet->codecSpecificHeader.codecHeader.VP8.layerSync = true;
frame.InsertPacket(*packet, 0, false, 0);
EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
EXPECT_TRUE(dec_state.full_sync());
delete packet;
}
TEST(TestDecodingState, DiscontinuousPicIdContinuousSeqNum) {
VCMDecodingState dec_state;
VCMFrameBuffer frame;
VCMPacket packet;
frame.Reset();
frame.SetState(kStateEmpty);
packet.frameType = kVideoFrameKey;
packet.codecSpecificHeader.codec = kRTPVideoVP8;
packet.timestamp = 0;
packet.seqNum = 0;
packet.codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
packet.codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
packet.codecSpecificHeader.codecHeader.VP8.pictureId = 0;
frame.InsertPacket(packet, 0, false, 0);
dec_state.SetState(&frame);
EXPECT_TRUE(dec_state.full_sync());
// Continuous sequence number but discontinuous picture id. This implies a
// a loss and we have to fall back to only decoding the base layer.
frame.Reset();
frame.SetState(kStateEmpty);
packet.frameType = kVideoFrameDelta;
packet.timestamp += 3000;
++packet.seqNum;
packet.codecSpecificHeader.codecHeader.VP8.temporalIdx = 1;
packet.codecSpecificHeader.codecHeader.VP8.pictureId = 2;
frame.InsertPacket(packet, 0, false, 0);
EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
dec_state.SetState(&frame);
EXPECT_FALSE(dec_state.full_sync());
}
TEST(TestDecodingState, OldInput) {
VCMDecodingState dec_state;
// Identify packets belonging to old frames/packets.
// Set state for current frames.
VCMFrameBuffer frame;
frame.SetState(kStateEmpty);
VCMPacket* packet = new VCMPacket();
packet->timestamp = 10;
packet->seqNum = 1;
frame.InsertPacket(*packet, 0, false, 0);
dec_state.SetState(&frame);
packet->timestamp = 9;
EXPECT_TRUE(dec_state.IsOldPacket(packet));
// Check for old frame
frame.Reset();
frame.InsertPacket(*packet, 0, false, 0);
EXPECT_TRUE(dec_state.IsOldFrame(&frame));
delete packet;
}
} // namespace webrtc

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "encoded_frame.h"
#include "generic_encoder.h"
#include "jitter_buffer_common.h"
#include "video_coding_defines.h"
namespace webrtc {
VCMEncodedFrame::VCMEncodedFrame()
:
webrtc::EncodedImage(),
_renderTimeMs(-1),
_payloadType(0),
_missingFrame(false),
_codec(kVideoCodecUnknown),
_fragmentation()
{
_codecSpecificInfo.codecType = kVideoCodecUnknown;
}
VCMEncodedFrame::VCMEncodedFrame(const webrtc::EncodedImage& rhs)
:
webrtc::EncodedImage(rhs),
_renderTimeMs(-1),
_payloadType(0),
_missingFrame(false),
_codec(kVideoCodecUnknown),
_fragmentation()
{
_codecSpecificInfo.codecType = kVideoCodecUnknown;
_buffer = NULL;
_size = 0;
_length = 0;
if (rhs._buffer != NULL)
{
VerifyAndAllocate(rhs._length);
memcpy(_buffer, rhs._buffer, rhs._length);
}
}
VCMEncodedFrame::VCMEncodedFrame(const VCMEncodedFrame& rhs)
:
webrtc::EncodedImage(rhs),
_renderTimeMs(rhs._renderTimeMs),
_payloadType(rhs._payloadType),
_missingFrame(rhs._missingFrame),
_codecSpecificInfo(rhs._codecSpecificInfo),
_codec(rhs._codec),
_fragmentation() {
_buffer = NULL;
_size = 0;
_length = 0;
if (rhs._buffer != NULL)
{
VerifyAndAllocate(rhs._length);
memcpy(_buffer, rhs._buffer, rhs._length);
_length = rhs._length;
}
// Deep operator=
_fragmentation = rhs._fragmentation;
}
VCMEncodedFrame::~VCMEncodedFrame()
{
Free();
}
void VCMEncodedFrame::Free()
{
Reset();
if (_buffer != NULL)
{
delete [] _buffer;
_buffer = NULL;
}
}
void VCMEncodedFrame::Reset()
{
_renderTimeMs = -1;
_timeStamp = 0;
_payloadType = 0;
_frameType = kDeltaFrame;
_encodedWidth = 0;
_encodedHeight = 0;
_completeFrame = false;
_missingFrame = false;
_length = 0;
_codecSpecificInfo.codecType = kVideoCodecUnknown;
_codec = kVideoCodecUnknown;
}
void VCMEncodedFrame::CopyCodecSpecific(const RTPVideoHeader* header)
{
if (header)
{
switch (header->codec)
{
case kRTPVideoVP8:
{
if (_codecSpecificInfo.codecType != kVideoCodecVP8)
{
// This is the first packet for this frame.
_codecSpecificInfo.codecSpecific.VP8.pictureId = -1;
_codecSpecificInfo.codecSpecific.VP8.temporalIdx = 0;
_codecSpecificInfo.codecSpecific.VP8.layerSync = false;
_codecSpecificInfo.codecSpecific.VP8.keyIdx = -1;
_codecSpecificInfo.codecType = kVideoCodecVP8;
}
_codecSpecificInfo.codecSpecific.VP8.nonReference =
header->codecHeader.VP8.nonReference;
if (header->codecHeader.VP8.pictureId != kNoPictureId)
{
_codecSpecificInfo.codecSpecific.VP8.pictureId =
header->codecHeader.VP8.pictureId;
}
if (header->codecHeader.VP8.temporalIdx != kNoTemporalIdx)
{
_codecSpecificInfo.codecSpecific.VP8.temporalIdx =
header->codecHeader.VP8.temporalIdx;
_codecSpecificInfo.codecSpecific.VP8.layerSync =
header->codecHeader.VP8.layerSync;
}
if (header->codecHeader.VP8.keyIdx != kNoKeyIdx)
{
_codecSpecificInfo.codecSpecific.VP8.keyIdx =
header->codecHeader.VP8.keyIdx;
}
break;
}
default:
{
_codecSpecificInfo.codecType = kVideoCodecUnknown;
break;
}
}
}
}
const RTPFragmentationHeader* VCMEncodedFrame::FragmentationHeader() const {
return &_fragmentation;
}
WebRtc_Word32
VCMEncodedFrame::Store(VCMFrameStorageCallback& storeCallback) const
{
EncodedVideoData frameToStore;
frameToStore.codec = _codec;
if (_buffer != NULL)
{
frameToStore.VerifyAndAllocate(_length);
memcpy(frameToStore.payloadData, _buffer, _length);
frameToStore.payloadSize = _length;
}
frameToStore.completeFrame = _completeFrame;
frameToStore.encodedWidth = _encodedWidth;
frameToStore.encodedHeight = _encodedHeight;
frameToStore.frameType = ConvertFrameType(_frameType);
frameToStore.missingFrame = _missingFrame;
frameToStore.payloadType = _payloadType;
frameToStore.renderTimeMs = _renderTimeMs;
frameToStore.timeStamp = _timeStamp;
storeCallback.StoreReceivedFrame(frameToStore);
return VCM_OK;
}
WebRtc_Word32
VCMEncodedFrame::VerifyAndAllocate(const WebRtc_UWord32 minimumSize)
{
if(minimumSize > _size)
{
// create buffer of sufficient size
WebRtc_UWord8* newBuffer = new WebRtc_UWord8[minimumSize];
if (newBuffer == NULL)
{
return -1;
}
if(_buffer)
{
// copy old data
memcpy(newBuffer, _buffer, _size);
delete [] _buffer;
}
_buffer = newBuffer;
_size = minimumSize;
}
return 0;
}
webrtc::FrameType VCMEncodedFrame::ConvertFrameType(VideoFrameType frameType)
{
switch(frameType)
{
case kKeyFrame:
{
return kVideoFrameKey;
}
case kDeltaFrame:
{
return kVideoFrameDelta;
}
case kGoldenFrame:
{
return kVideoFrameGolden;
}
case kAltRefFrame:
{
return kVideoFrameAltRef;
}
case kSkipFrame:
{
return kFrameEmpty;
}
default:
{
return kVideoFrameDelta;
}
}
}
VideoFrameType VCMEncodedFrame::ConvertFrameType(webrtc::FrameType frame_type) {
switch (frame_type) {
case kVideoFrameKey:
return kKeyFrame;
case kVideoFrameDelta:
return kDeltaFrame;
case kVideoFrameGolden:
return kGoldenFrame;
case kVideoFrameAltRef:
return kAltRefFrame;
default:
assert(false);
return kDeltaFrame;
}
}
void VCMEncodedFrame::ConvertFrameTypes(
const std::vector<webrtc::FrameType>& frame_types,
std::vector<VideoFrameType>* video_frame_types) {
assert(video_frame_types);
video_frame_types->reserve(frame_types.size());
for (size_t i = 0; i < frame_types.size(); ++i) {
(*video_frame_types)[i] = ConvertFrameType(frame_types[i]);
}
}
}

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_ENCODED_FRAME_H_
#define WEBRTC_MODULES_VIDEO_CODING_ENCODED_FRAME_H_
#include <vector>
#include "common_types.h"
#include "common_video/interface/video_image.h"
#include "modules/interface/module_common_types.h"
#include "modules/video_coding/codecs/interface/video_codec_interface.h"
#include "modules/video_coding/main/interface/video_coding_defines.h"
namespace webrtc
{
class VCMEncodedFrame : protected EncodedImage
{
public:
VCMEncodedFrame();
VCMEncodedFrame(const webrtc::EncodedImage& rhs);
VCMEncodedFrame(const VCMEncodedFrame& rhs);
~VCMEncodedFrame();
/**
* Delete VideoFrame and resets members to zero
*/
void Free();
/**
* Set render time in milliseconds
*/
void SetRenderTime(const WebRtc_Word64 renderTimeMs) {_renderTimeMs = renderTimeMs;}
/**
* Set the encoded frame size
*/
void SetEncodedSize(WebRtc_UWord32 width, WebRtc_UWord32 height)
{ _encodedWidth = width; _encodedHeight = height; }
/**
* Get the encoded image
*/
const webrtc::EncodedImage& EncodedImage() const
{ return static_cast<const webrtc::EncodedImage&>(*this); }
/**
* Get pointer to frame buffer
*/
const WebRtc_UWord8* Buffer() const {return _buffer;}
/**
* Get frame length
*/
WebRtc_UWord32 Length() const {return _length;}
/**
* Get frame timestamp (90kHz)
*/
WebRtc_UWord32 TimeStamp() const {return _timeStamp;}
/**
* Get render time in milliseconds
*/
WebRtc_Word64 RenderTimeMs() const {return _renderTimeMs;}
/**
* Get frame type
*/
webrtc::FrameType FrameType() const {return ConvertFrameType(_frameType);}
/**
* True if this frame is complete, false otherwise
*/
bool Complete() const { return _completeFrame; }
/**
* True if there's a frame missing before this frame
*/
bool MissingFrame() const { return _missingFrame; }
/**
* Payload type of the encoded payload
*/
WebRtc_UWord8 PayloadType() const { return _payloadType; }
/**
* Get codec specific info.
* The returned pointer is only valid as long as the VCMEncodedFrame
* is valid. Also, VCMEncodedFrame owns the pointer and will delete
* the object.
*/
const CodecSpecificInfo* CodecSpecific() const {return &_codecSpecificInfo;}
const RTPFragmentationHeader* FragmentationHeader() const;
WebRtc_Word32 Store(VCMFrameStorageCallback& storeCallback) const;
static webrtc::FrameType ConvertFrameType(VideoFrameType frameType);
static VideoFrameType ConvertFrameType(webrtc::FrameType frameType);
static void ConvertFrameTypes(
const std::vector<webrtc::FrameType>& frame_types,
std::vector<VideoFrameType>* video_frame_types);
protected:
/**
* Verifies that current allocated buffer size is larger than or equal to the input size.
* If the current buffer size is smaller, a new allocation is made and the old buffer data
* is copied to the new buffer.
* Buffer size is updated to minimumSize.
*/
WebRtc_Word32 VerifyAndAllocate(const WebRtc_UWord32 minimumSize);
void Reset();
void CopyCodecSpecific(const RTPVideoHeader* header);
WebRtc_Word64 _renderTimeMs;
WebRtc_UWord8 _payloadType;
bool _missingFrame;
CodecSpecificInfo _codecSpecificInfo;
webrtc::VideoCodecType _codec;
RTPFragmentationHeader _fragmentation;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_ENCODED_FRAME_H_

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_EVENT_H_
#define WEBRTC_MODULES_VIDEO_CODING_EVENT_H_
#include "event_wrapper.h"
namespace webrtc
{
//#define EVENT_DEBUG
class VCMEvent : public EventWrapper
{
public:
VCMEvent() : _event(*EventWrapper::Create()) {};
virtual ~VCMEvent() { delete &_event; };
/**
* Release waiting threads
*/
bool Set() { return _event.Set(); };
bool Reset() { return _event.Reset(); };
/**
* Wait for this event
*/
EventTypeWrapper Wait(unsigned long maxTime)
{
#ifdef EVENT_DEBUG
return kEventTimeout;
#else
return _event.Wait(maxTime);
#endif
};
/**
* Start a timer
*/
bool StartTimer(bool periodic, unsigned long time)
{ return _event.StartTimer(periodic, time); };
/**
* Stop the timer
*/
bool StopTimer() { return _event.StopTimer(); };
private:
EventWrapper& _event;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_EVENT_H_

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "exp_filter.h"
#include <math.h>
namespace webrtc {
void
VCMExpFilter::Reset(float alpha)
{
_alpha = alpha;
_filtered = -1.0;
}
float
VCMExpFilter::Apply(float exp, float sample)
{
if (_filtered == -1.0)
{
// Initialize filtered bit rates
_filtered = sample;
}
else if (exp == 1.0)
{
_filtered = _alpha * _filtered + (1 - _alpha) * sample;
}
else
{
float alpha = pow(_alpha, exp);
_filtered = alpha * _filtered + (1 - alpha) * sample;
}
if (_max != -1 && _filtered > _max)
{
_filtered = _max;
}
return _filtered;
}
void
VCMExpFilter::UpdateBase(float alpha)
{
_alpha = alpha;
}
float
VCMExpFilter::Value() const
{
return _filtered;
}
}

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_EXP_FILTER_H_
#define WEBRTC_MODULES_VIDEO_CODING_EXP_FILTER_H_
namespace webrtc
{
/**********************/
/* ExpFilter class */
/**********************/
class VCMExpFilter
{
public:
VCMExpFilter(float alpha, float max = -1.0) : _alpha(alpha), _filtered(-1.0), _max(max) {}
// Resets the filter to its initial state, and resets alpha to the given value
//
// Input:
// - alpha : the new value of the filter factor base.
void Reset(float alpha);
// Applies the filter with the given exponent on the provided sample
//
// Input:
// - exp : Exponent T in y(k) = alpha^T * y(k-1) + (1 - alpha^T) * x(k)
// - sample : x(k) in the above filter equation
float Apply(float exp, float sample);
// Return current filtered value: y(k)
//
// Return value : The current filter output
float Value() const;
// Change the filter factor base
//
// Input:
// - alpha : The new filter factor base.
void UpdateBase(float alpha);
private:
float _alpha; // Filter factor base
float _filtered; // Current filter output
const float _max;
}; // end of ExpFilter class
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_EXP_FILTER_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "frame_buffer.h"
#include "packet.h"
#include <cassert>
#include <string.h>
namespace webrtc {
VCMFrameBuffer::VCMFrameBuffer()
:
_state(kStateFree),
_frameCounted(false),
_nackCount(0),
_latestPacketTimeMs(-1) {
}
VCMFrameBuffer::~VCMFrameBuffer() {
}
VCMFrameBuffer::VCMFrameBuffer(VCMFrameBuffer& rhs)
:
VCMEncodedFrame(rhs),
_state(rhs._state),
_frameCounted(rhs._frameCounted),
_sessionInfo(),
_nackCount(rhs._nackCount),
_latestPacketTimeMs(rhs._latestPacketTimeMs)
{
_sessionInfo = rhs._sessionInfo;
_sessionInfo.UpdateDataPointers(rhs._buffer, _buffer);
}
webrtc::FrameType
VCMFrameBuffer::FrameType() const
{
return _sessionInfo.FrameType();
}
void
VCMFrameBuffer::SetPreviousFrameLoss()
{
_sessionInfo.SetPreviousFrameLoss();
}
WebRtc_Word32
VCMFrameBuffer::GetLowSeqNum() const
{
return _sessionInfo.LowSequenceNumber();
}
WebRtc_Word32
VCMFrameBuffer::GetHighSeqNum() const
{
return _sessionInfo.HighSequenceNumber();
}
int VCMFrameBuffer::PictureId() const {
return _sessionInfo.PictureId();
}
int VCMFrameBuffer::TemporalId() const {
return _sessionInfo.TemporalId();
}
bool VCMFrameBuffer::LayerSync() const {
return _sessionInfo.LayerSync();
}
int VCMFrameBuffer::Tl0PicId() const {
return _sessionInfo.Tl0PicId();
}
bool VCMFrameBuffer::NonReference() const {
return _sessionInfo.NonReference();
}
bool
VCMFrameBuffer::IsSessionComplete() const
{
return _sessionInfo.complete();
}
// Insert packet
VCMFrameBufferEnum
VCMFrameBuffer::InsertPacket(const VCMPacket& packet, WebRtc_Word64 timeInMs,
bool enableDecodableState, WebRtc_UWord32 rttMS)
{
if (_state == kStateDecoding)
{
// Do not insert packet
return kNoError;
}
// Sanity to check if the frame has been freed. (Too old for example)
if (_state == kStateFree)
{
return kStateError;
}
// is this packet part of this frame
if (TimeStamp() && (TimeStamp() != packet.timestamp))
{
return kTimeStampError;
}
// sanity checks
if (_size + packet.sizeBytes +
(packet.insertStartCode ? kH264StartCodeLengthBytes : 0 )
> kMaxJBFrameSizeBytes)
{
return kSizeError;
}
if (NULL == packet.dataPtr && packet.sizeBytes > 0)
{
return kSizeError;
}
if (packet.dataPtr != NULL)
{
_payloadType = packet.payloadType;
}
if (kStateEmpty == _state)
{
// First packet (empty and/or media) inserted into this frame.
// store some info and set some initial values.
_timeStamp = packet.timestamp;
_codec = packet.codec;
if (packet.frameType != kFrameEmpty)
{
// first media packet
SetState(kStateIncomplete);
}
}
WebRtc_UWord32 requiredSizeBytes = Length() + packet.sizeBytes +
(packet.insertStartCode ? kH264StartCodeLengthBytes : 0);
if (requiredSizeBytes >= _size)
{
const WebRtc_UWord8* prevBuffer = _buffer;
const WebRtc_UWord32 increments = requiredSizeBytes /
kBufferIncStepSizeBytes +
(requiredSizeBytes %
kBufferIncStepSizeBytes > 0);
const WebRtc_UWord32 newSize = _size +
increments * kBufferIncStepSizeBytes;
if (newSize > kMaxJBFrameSizeBytes)
{
return kSizeError;
}
if (VerifyAndAllocate(newSize) == -1)
{
return kSizeError;
}
_sessionInfo.UpdateDataPointers(prevBuffer, _buffer);
}
CopyCodecSpecific(&packet.codecSpecificHeader);
int retVal = _sessionInfo.InsertPacket(packet, _buffer,
enableDecodableState,
rttMS);
if (retVal == -1)
{
return kSizeError;
}
else if (retVal == -2)
{
return kDuplicatePacket;
}
// update length
_length = Length() + static_cast<WebRtc_UWord32>(retVal);
_latestPacketTimeMs = timeInMs;
if (_sessionInfo.complete()) {
return kCompleteSession;
} else if (_sessionInfo.decodable()) {
SetState(kStateDecodable);
return kDecodableSession;
} else {
// this layer is not complete
if (_state == kStateComplete) {
// we already have a complete layer
// wait for all independent layers belonging to the same frame
_state = kStateIncomplete;
}
}
return kIncomplete;
}
WebRtc_Word64
VCMFrameBuffer::LatestPacketTimeMs() const
{
return _latestPacketTimeMs;
}
// Build hard NACK list:Zero out all entries in list up to and including the
// (first) entry equal to _lowSeqNum.
int VCMFrameBuffer::BuildHardNackList(int* list, int num) {
if (_sessionInfo.BuildHardNackList(list, num) != 0) {
return -1;
}
return 0;
}
// Build selective NACK list: Create a soft (selective) list of entries to zero
// out up to and including the (first) entry equal to _lowSeqNum.
int VCMFrameBuffer::BuildSoftNackList(int* list, int num, int rttMs) {
return _sessionInfo.BuildSoftNackList(list, num, rttMs);
}
void
VCMFrameBuffer::IncrementNackCount()
{
_nackCount++;
}
WebRtc_Word16
VCMFrameBuffer::GetNackCount() const
{
return _nackCount;
}
bool
VCMFrameBuffer::HaveLastPacket() const
{
return _sessionInfo.HaveLastPacket();
}
void
VCMFrameBuffer::Reset()
{
_length = 0;
_timeStamp = 0;
_sessionInfo.Reset();
_frameCounted = false;
_payloadType = 0;
_nackCount = 0;
_latestPacketTimeMs = -1;
_state = kStateFree;
VCMEncodedFrame::Reset();
}
// Makes sure the session contains a decodable stream.
void
VCMFrameBuffer::MakeSessionDecodable()
{
WebRtc_UWord32 retVal;
#ifdef INDEPENDENT_PARTITIONS
if (_codec != kVideoCodecVP8) {
retVal = _sessionInfo.MakeDecodable();
_length -= retVal;
}
#else
retVal = _sessionInfo.MakeDecodable();
_length -= retVal;
#endif
}
// Set state of frame
void
VCMFrameBuffer::SetState(VCMFrameBufferStateEnum state)
{
if (_state == state)
{
return;
}
switch (state)
{
case kStateFree:
// Reset everything
// We can go to this state from all other states.
// The one setting the state to free must ensure
// that the frame is removed from the timestamp
// ordered frame list in the jb.
Reset();
break;
case kStateIncomplete:
// we can go to this state from state kStateEmpty
assert(_state == kStateEmpty ||
_state == kStateDecoding);
// Do nothing, we received a packet
break;
case kStateComplete:
assert(_state == kStateEmpty ||
_state == kStateIncomplete ||
_state == kStateDecodable);
break;
case kStateEmpty:
assert(_state == kStateFree);
// Do nothing
break;
case kStateDecoding:
// A frame might have received empty packets, or media packets might
// have been removed when making the frame decodable. The frame can
// still be set to decodable since it can be used to inform the
// decoder of a frame loss.
assert(_state == kStateComplete || _state == kStateIncomplete ||
_state == kStateDecodable || _state == kStateEmpty);
// Transfer frame information to EncodedFrame and create any codec
// specific information
RestructureFrameInformation();
break;
case kStateDecodable:
assert(_state == kStateEmpty ||
_state == kStateIncomplete);
break;
}
_state = state;
}
void
VCMFrameBuffer::RestructureFrameInformation()
{
PrepareForDecode();
_frameType = ConvertFrameType(_sessionInfo.FrameType());
_completeFrame = _sessionInfo.complete();
_missingFrame = _sessionInfo.PreviousFrameLoss();
}
WebRtc_Word32
VCMFrameBuffer::ExtractFromStorage(const EncodedVideoData& frameFromStorage)
{
_frameType = ConvertFrameType(frameFromStorage.frameType);
_timeStamp = frameFromStorage.timeStamp;
_payloadType = frameFromStorage.payloadType;
_encodedWidth = frameFromStorage.encodedWidth;
_encodedHeight = frameFromStorage.encodedHeight;
_missingFrame = frameFromStorage.missingFrame;
_completeFrame = frameFromStorage.completeFrame;
_renderTimeMs = frameFromStorage.renderTimeMs;
_codec = frameFromStorage.codec;
const WebRtc_UWord8 *prevBuffer = _buffer;
if (VerifyAndAllocate(frameFromStorage.payloadSize) < 0)
{
return VCM_MEMORY;
}
_sessionInfo.UpdateDataPointers(prevBuffer, _buffer);
memcpy(_buffer, frameFromStorage.payloadData, frameFromStorage.payloadSize);
_length = frameFromStorage.payloadSize;
return VCM_OK;
}
int VCMFrameBuffer::NotDecodablePackets() const {
return _sessionInfo.packets_not_decodable();
}
// Set counted status (as counted by JB or not)
void VCMFrameBuffer::SetCountedFrame(bool frameCounted)
{
_frameCounted = frameCounted;
}
bool VCMFrameBuffer::GetCountedFrame() const
{
return _frameCounted;
}
// Get current state of frame
VCMFrameBufferStateEnum
VCMFrameBuffer::GetState() const
{
return _state;
}
// Get current state of frame
VCMFrameBufferStateEnum
VCMFrameBuffer::GetState(WebRtc_UWord32& timeStamp) const
{
timeStamp = TimeStamp();
return GetState();
}
bool
VCMFrameBuffer::IsRetransmitted() const
{
return _sessionInfo.session_nack();
}
void
VCMFrameBuffer::PrepareForDecode()
{
#ifdef INDEPENDENT_PARTITIONS
if (_codec == kVideoCodecVP8)
{
_length =
_sessionInfo.BuildVP8FragmentationHeader(_buffer, _length,
&_fragmentation);
}
#endif
}
}

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_FRAME_BUFFER_H_
#define WEBRTC_MODULES_VIDEO_CODING_FRAME_BUFFER_H_
#include "modules/interface/module_common_types.h"
#include "modules/video_coding/main/source/encoded_frame.h"
#include "modules/video_coding/main/source/jitter_buffer_common.h"
#include "modules/video_coding/main/source/session_info.h"
#include "typedefs.h"
namespace webrtc
{
class VCMFrameBuffer : public VCMEncodedFrame
{
public:
VCMFrameBuffer();
virtual ~VCMFrameBuffer();
VCMFrameBuffer(VCMFrameBuffer& rhs);
virtual void Reset();
VCMFrameBufferEnum InsertPacket(const VCMPacket& packet,
WebRtc_Word64 timeInMs,
bool enableDecodableState,
WebRtc_UWord32 rttMs);
// State
// Get current state of frame
VCMFrameBufferStateEnum GetState() const;
// Get current state and timestamp of frame
VCMFrameBufferStateEnum GetState(WebRtc_UWord32& timeStamp) const;
void SetState(VCMFrameBufferStateEnum state); // Set state of frame
bool IsRetransmitted() const;
bool IsSessionComplete() const;
bool HaveLastPacket() const;
// Makes sure the session contain a decodable stream.
void MakeSessionDecodable();
// Sequence numbers
// Get lowest packet sequence number in frame
WebRtc_Word32 GetLowSeqNum() const;
// Get highest packet sequence number in frame
WebRtc_Word32 GetHighSeqNum() const;
int PictureId() const;
int TemporalId() const;
bool LayerSync() const;
int Tl0PicId() const;
bool NonReference() const;
// Set counted status (as counted by JB or not)
void SetCountedFrame(bool frameCounted);
bool GetCountedFrame() const;
// NACK - Building the NACK lists.
// Build hard NACK list: Zero out all entries in list up to and including
// _lowSeqNum.
int BuildHardNackList(int* list, int num);
// Build soft NACK list: Zero out only a subset of the packets, discard
// empty packets.
int BuildSoftNackList(int* list, int num, int rttMs);
void IncrementNackCount();
WebRtc_Word16 GetNackCount() const;
WebRtc_Word64 LatestPacketTimeMs() const;
webrtc::FrameType FrameType() const;
void SetPreviousFrameLoss();
WebRtc_Word32 ExtractFromStorage(const EncodedVideoData& frameFromStorage);
// The number of packets discarded because the decoder can't make use of
// them.
int NotDecodablePackets() const;
protected:
void RestructureFrameInformation();
void PrepareForDecode();
private:
VCMFrameBufferStateEnum _state; // Current state of the frame
bool _frameCounted; // Was this frame counted by JB?
VCMSessionInfo _sessionInfo;
WebRtc_UWord16 _nackCount;
WebRtc_Word64 _latestPacketTimeMs;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_FRAME_BUFFER_H_

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "frame_dropper.h"
#include "internal_defines.h"
#include "trace.h"
namespace webrtc
{
VCMFrameDropper::VCMFrameDropper(WebRtc_Word32 vcmId)
:
_vcmId(vcmId),
_keyFrameSizeAvgKbits(0.9f),
_keyFrameRatio(0.99f),
_dropRatio(0.9f, 0.96f)
{
Reset();
}
void
VCMFrameDropper::Reset()
{
_keyFrameRatio.Reset(0.99f);
_keyFrameRatio.Apply(1.0f, 1.0f/300.0f); // 1 key frame every 10th second in 30 fps
_keyFrameSizeAvgKbits.Reset(0.9f);
_keyFrameCount = 0;
_accumulator = 0.0f;
_accumulatorMax = 150.0f; // assume 300 kb/s and 0.5 s window
_targetBitRate = 300.0f;
_incoming_frame_rate = 30;
_keyFrameSpreadFrames = 0.5f * _incoming_frame_rate;
_dropNext = false;
_dropRatio.Reset(0.9f);
_dropRatio.Apply(0.0f, 0.0f); // Initialize to 0
_dropCount = 0;
_windowSize = 0.5f;
_wasBelowMax = true;
_enabled = true;
_fastMode = false; // start with normal (non-aggressive) mode
// Cap for the encoder buffer level/accumulator, in secs.
_cap_buffer_size = 3.0f;
// Cap on maximum amount of dropped frames between kept frames, in secs.
_max_time_drops = 4.0f;
}
void
VCMFrameDropper::Enable(bool enable)
{
_enabled = enable;
}
void
VCMFrameDropper::Fill(WebRtc_UWord32 frameSizeBytes, bool deltaFrame)
{
if (!_enabled)
{
return;
}
float frameSizeKbits = 8.0f * static_cast<float>(frameSizeBytes) / 1000.0f;
if (!deltaFrame && !_fastMode) // fast mode does not treat key-frames any different
{
_keyFrameSizeAvgKbits.Apply(1, frameSizeKbits);
_keyFrameRatio.Apply(1.0, 1.0);
if (frameSizeKbits > _keyFrameSizeAvgKbits.Value())
{
// Remove the average key frame size since we
// compensate for key frames when adding delta
// frames.
frameSizeKbits -= _keyFrameSizeAvgKbits.Value();
}
else
{
// Shouldn't be negative, so zero is the lower bound.
frameSizeKbits = 0;
}
if (_keyFrameRatio.Value() > 1e-5 && 1 / _keyFrameRatio.Value() < _keyFrameSpreadFrames)
{
// We are sending key frames more often than our upper bound for
// how much we allow the key frame compensation to be spread
// out in time. Therefor we must use the key frame ratio rather
// than keyFrameSpreadFrames.
_keyFrameCount = static_cast<WebRtc_Word32>(1 / _keyFrameRatio.Value() + 0.5);
}
else
{
// Compensate for the key frame the following frames
_keyFrameCount = static_cast<WebRtc_Word32>(_keyFrameSpreadFrames + 0.5);
}
}
else
{
// Decrease the keyFrameRatio
_keyFrameRatio.Apply(1.0, 0.0);
}
// Change the level of the accumulator (bucket)
_accumulator += frameSizeKbits;
CapAccumulator();
}
void
VCMFrameDropper::Leak(WebRtc_UWord32 inputFrameRate)
{
if (!_enabled)
{
return;
}
if (inputFrameRate < 1)
{
return;
}
if (_targetBitRate < 0.0f)
{
return;
}
_keyFrameSpreadFrames = 0.5f * inputFrameRate;
// T is the expected bits per frame (target). If all frames were the same size,
// we would get T bits per frame. Notice that T is also weighted to be able to
// force a lower frame rate if wanted.
float T = _targetBitRate / inputFrameRate;
if (_keyFrameCount > 0)
{
// Perform the key frame compensation
if (_keyFrameRatio.Value() > 0 && 1 / _keyFrameRatio.Value() < _keyFrameSpreadFrames)
{
T -= _keyFrameSizeAvgKbits.Value() * _keyFrameRatio.Value();
}
else
{
T -= _keyFrameSizeAvgKbits.Value() / _keyFrameSpreadFrames;
}
_keyFrameCount--;
}
_accumulator -= T;
UpdateRatio();
}
void
VCMFrameDropper::UpdateNack(WebRtc_UWord32 nackBytes)
{
if (!_enabled)
{
return;
}
_accumulator += static_cast<float>(nackBytes) * 8.0f / 1000.0f;
}
void
VCMFrameDropper::FillBucket(float inKbits, float outKbits)
{
_accumulator += (inKbits - outKbits);
}
void
VCMFrameDropper::UpdateRatio()
{
if (_accumulator > 1.3f * _accumulatorMax)
{
// Too far above accumulator max, react faster
_dropRatio.UpdateBase(0.8f);
}
else
{
// Go back to normal reaction
_dropRatio.UpdateBase(0.9f);
}
if (_accumulator > _accumulatorMax)
{
// We are above accumulator max, and should ideally
// drop a frame. Increase the dropRatio and drop
// the frame later.
if (_wasBelowMax)
{
_dropNext = true;
}
if (_fastMode)
{
// always drop in aggressive mode
_dropNext = true;
}
_dropRatio.Apply(1.0f, 1.0f);
_dropRatio.UpdateBase(0.9f);
}
else
{
_dropRatio.Apply(1.0f, 0.0f);
}
if (_accumulator < 0.0f)
{
_accumulator = 0.0f;
}
_wasBelowMax = _accumulator < _accumulatorMax;
WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId), "FrameDropper: dropRatio = %f accumulator = %f, accumulatorMax = %f", _dropRatio.Value(), _accumulator, _accumulatorMax);
}
// This function signals when to drop frames to the caller. It makes use of the dropRatio
// to smooth out the drops over time.
bool
VCMFrameDropper::DropFrame()
{
if (!_enabled)
{
return false;
}
if (_dropNext)
{
_dropNext = false;
_dropCount = 0;
}
if (_dropRatio.Value() >= 0.5f) // Drops per keep
{
// limit is the number of frames we should drop between each kept frame
// to keep our drop ratio. limit is positive in this case.
float denom = 1.0f - _dropRatio.Value();
if (denom < 1e-5)
{
denom = (float)1e-5;
}
WebRtc_Word32 limit = static_cast<WebRtc_Word32>(1.0f / denom - 1.0f + 0.5f);
// Put a bound on the max amount of dropped frames between each kept
// frame, in terms of frame rate and window size (secs).
int max_limit = static_cast<int>(_incoming_frame_rate *
_max_time_drops);
if (limit > max_limit) {
limit = max_limit;
}
if (_dropCount < 0)
{
// Reset the _dropCount since it was negative and should be positive.
if (_dropRatio.Value() > 0.4f)
{
_dropCount = -_dropCount;
}
else
{
_dropCount = 0;
}
}
if (_dropCount < limit)
{
// As long we are below the limit we should drop frames.
_dropCount++;
return true;
}
else
{
// Only when we reset _dropCount a frame should be kept.
_dropCount = 0;
return false;
}
}
else if (_dropRatio.Value() > 0.0f && _dropRatio.Value() < 0.5f) // Keeps per drop
{
// limit is the number of frames we should keep between each drop
// in order to keep the drop ratio. limit is negative in this case,
// and the _dropCount is also negative.
float denom = _dropRatio.Value();
if (denom < 1e-5)
{
denom = (float)1e-5;
}
WebRtc_Word32 limit = -static_cast<WebRtc_Word32>(1.0f / denom - 1.0f + 0.5f);
if (_dropCount > 0)
{
// Reset the _dropCount since we have a positive
// _dropCount, and it should be negative.
if (_dropRatio.Value() < 0.6f)
{
_dropCount = -_dropCount;
}
else
{
_dropCount = 0;
}
}
if (_dropCount > limit)
{
if (_dropCount == 0)
{
// Drop frames when we reset _dropCount.
_dropCount--;
return true;
}
else
{
// Keep frames as long as we haven't reached limit.
_dropCount--;
return false;
}
}
else
{
_dropCount = 0;
return false;
}
}
_dropCount = 0;
return false;
// A simpler version, unfiltered and quicker
//bool dropNext = _dropNext;
//_dropNext = false;
//return dropNext;
}
void
VCMFrameDropper::SetRates(float bitRate, float incoming_frame_rate)
{
// Bit rate of -1 means infinite bandwidth.
_accumulatorMax = bitRate * _windowSize; // bitRate * windowSize (in seconds)
if (_targetBitRate > 0.0f && bitRate < _targetBitRate && _accumulator > _accumulatorMax)
{
// Rescale the accumulator level if the accumulator max decreases
_accumulator = bitRate / _targetBitRate * _accumulator;
}
_targetBitRate = bitRate;
CapAccumulator();
_incoming_frame_rate = incoming_frame_rate;
}
float
VCMFrameDropper::ActualFrameRate(WebRtc_UWord32 inputFrameRate) const
{
if (!_enabled)
{
return static_cast<float>(inputFrameRate);
}
return inputFrameRate * (1.0f - _dropRatio.Value());
}
// Put a cap on the accumulator, i.e., don't let it grow beyond some level.
// This is a temporary fix for screencasting where very large frames from
// encoder will cause very slow response (too many frame drops).
void VCMFrameDropper::CapAccumulator() {
float max_accumulator = _targetBitRate * _cap_buffer_size;
if (_accumulator > max_accumulator) {
_accumulator = max_accumulator;
}
}
}

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_FRAME_DROPPER_H_
#define WEBRTC_MODULES_VIDEO_CODING_FRAME_DROPPER_H_
#include "exp_filter.h"
#include "typedefs.h"
namespace webrtc
{
/******************************/
/* VCMFrameDropper class */
/****************************/
// The Frame Dropper implements a variant of the leaky bucket algorithm
// for keeping track of when to drop frames to avoid bit rate
// over use when the encoder can't keep its bit rate.
class VCMFrameDropper
{
public:
VCMFrameDropper(WebRtc_Word32 vcmId = 0);
// Resets the FrameDropper to its initial state.
// This means that the frameRateWeight is set to its
// default value as well.
void Reset();
void Enable(bool enable);
// Answers the question if it's time to drop a frame
// if we want to reach a given frame rate. Must be
// called for every frame.
//
// Return value : True if we should drop the current frame
bool DropFrame();
// Updates the FrameDropper with the size of the latest encoded
// frame. The FrameDropper calculates a new drop ratio (can be
// seen as the probability to drop a frame) and updates its
// internal statistics.
//
// Input:
// - frameSizeBytes : The size of the latest frame
// returned from the encoder.
// - deltaFrame : True if the encoder returned
// a key frame.
void Fill(WebRtc_UWord32 frameSizeBytes, bool deltaFrame);
void Leak(WebRtc_UWord32 inputFrameRate);
void UpdateNack(WebRtc_UWord32 nackBytes);
// Sets the target bit rate and the frame rate produced by
// the camera.
//
// Input:
// - bitRate : The target bit rate
void SetRates(float bitRate, float incoming_frame_rate);
// Return value : The current average frame rate produced
// if the DropFrame() function is used as
// instruction of when to drop frames.
float ActualFrameRate(WebRtc_UWord32 inputFrameRate) const;
private:
void FillBucket(float inKbits, float outKbits);
void UpdateRatio();
void CapAccumulator();
WebRtc_Word32 _vcmId;
VCMExpFilter _keyFrameSizeAvgKbits;
VCMExpFilter _keyFrameRatio;
float _keyFrameSpreadFrames;
WebRtc_Word32 _keyFrameCount;
float _accumulator;
float _accumulatorMax;
float _targetBitRate;
bool _dropNext;
VCMExpFilter _dropRatio;
WebRtc_Word32 _dropCount;
float _windowSize;
float _incoming_frame_rate;
bool _wasBelowMax;
bool _enabled;
bool _fastMode;
float _cap_buffer_size;
float _max_time_drops;
}; // end of VCMFrameDropper class
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_FRAME_DROPPER_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "video_coding.h"
#include "trace.h"
#include "generic_decoder.h"
#include "internal_defines.h"
#include "tick_time_base.h"
namespace webrtc {
VCMDecodedFrameCallback::VCMDecodedFrameCallback(VCMTiming& timing,
TickTimeBase* clock)
:
_critSect(CriticalSectionWrapper::CreateCriticalSection()),
_clock(clock),
_receiveCallback(NULL),
_timing(timing),
_timestampMap(kDecoderFrameMemoryLength),
_lastReceivedPictureID(0)
{
}
VCMDecodedFrameCallback::~VCMDecodedFrameCallback()
{
delete _critSect;
}
void VCMDecodedFrameCallback::SetUserReceiveCallback(
VCMReceiveCallback* receiveCallback)
{
CriticalSectionScoped cs(_critSect);
_receiveCallback = receiveCallback;
}
WebRtc_Word32 VCMDecodedFrameCallback::Decoded(VideoFrame& decodedImage)
{
// TODO(holmer): We should improve this so that we can handle multiple
// callbacks from one call to Decode().
CriticalSectionScoped cs(_critSect);
VCMFrameInformation* frameInfo = static_cast<VCMFrameInformation*>(
_timestampMap.Pop(decodedImage.TimeStamp()));
if (frameInfo == NULL)
{
// The map should never be empty or full if this callback is called.
return WEBRTC_VIDEO_CODEC_ERROR;
}
_timing.StopDecodeTimer(
decodedImage.TimeStamp(),
frameInfo->decodeStartTimeMs,
_clock->MillisecondTimestamp());
if (_receiveCallback != NULL)
{
_frame.SwapFrame(decodedImage);
_frame.SetRenderTime(frameInfo->renderTimeMs);
WebRtc_Word32 callbackReturn = _receiveCallback->FrameToRender(_frame);
if (callbackReturn < 0)
{
WEBRTC_TRACE(webrtc::kTraceDebug,
webrtc::kTraceVideoCoding,
-1,
"Render callback returned error: %d", callbackReturn);
}
}
return WEBRTC_VIDEO_CODEC_OK;
}
WebRtc_Word32
VCMDecodedFrameCallback::ReceivedDecodedReferenceFrame(
const WebRtc_UWord64 pictureId)
{
CriticalSectionScoped cs(_critSect);
if (_receiveCallback != NULL)
{
return _receiveCallback->ReceivedDecodedReferenceFrame(pictureId);
}
return -1;
}
WebRtc_Word32
VCMDecodedFrameCallback::ReceivedDecodedFrame(const WebRtc_UWord64 pictureId)
{
_lastReceivedPictureID = pictureId;
return 0;
}
WebRtc_UWord64 VCMDecodedFrameCallback::LastReceivedPictureID() const
{
return _lastReceivedPictureID;
}
WebRtc_Word32 VCMDecodedFrameCallback::Map(WebRtc_UWord32 timestamp, VCMFrameInformation* frameInfo)
{
CriticalSectionScoped cs(_critSect);
return _timestampMap.Add(timestamp, frameInfo);
}
WebRtc_Word32 VCMDecodedFrameCallback::Pop(WebRtc_UWord32 timestamp)
{
CriticalSectionScoped cs(_critSect);
if (_timestampMap.Pop(timestamp) == NULL)
{
return VCM_GENERAL_ERROR;
}
return VCM_OK;
}
VCMGenericDecoder::VCMGenericDecoder(VideoDecoder& decoder, WebRtc_Word32 id, bool isExternal)
:
_id(id),
_callback(NULL),
_frameInfos(),
_nextFrameInfoIdx(0),
_decoder(decoder),
_codecType(kVideoCodecUnknown),
_isExternal(isExternal),
_requireKeyFrame(false),
_keyFrameDecoded(false)
{
}
VCMGenericDecoder::~VCMGenericDecoder()
{
}
WebRtc_Word32 VCMGenericDecoder::InitDecode(const VideoCodec* settings,
WebRtc_Word32 numberOfCores,
bool requireKeyFrame)
{
_requireKeyFrame = requireKeyFrame;
_keyFrameDecoded = false;
_codecType = settings->codecType;
return _decoder.InitDecode(settings, numberOfCores);
}
WebRtc_Word32 VCMGenericDecoder::Decode(const VCMEncodedFrame& frame,
int64_t nowMs)
{
if (_requireKeyFrame &&
!_keyFrameDecoded &&
frame.FrameType() != kVideoFrameKey &&
frame.FrameType() != kVideoFrameGolden)
{
// Require key frame is enabled, meaning that one key frame must be decoded
// before we can decode delta frames.
return VCM_CODEC_ERROR;
}
_frameInfos[_nextFrameInfoIdx].decodeStartTimeMs = nowMs;
_frameInfos[_nextFrameInfoIdx].renderTimeMs = frame.RenderTimeMs();
_callback->Map(frame.TimeStamp(), &_frameInfos[_nextFrameInfoIdx]);
WEBRTC_TRACE(webrtc::kTraceDebug,
webrtc::kTraceVideoCoding,
VCMId(_id),
"Decoding timestamp %u", frame.TimeStamp());
_nextFrameInfoIdx = (_nextFrameInfoIdx + 1) % kDecoderFrameMemoryLength;
WebRtc_Word32 ret = _decoder.Decode(frame.EncodedImage(),
frame.MissingFrame(),
frame.FragmentationHeader(),
frame.CodecSpecific(),
frame.RenderTimeMs());
if (ret < WEBRTC_VIDEO_CODEC_OK)
{
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding, VCMId(_id), "Decoder error: %d\n", ret);
_callback->Pop(frame.TimeStamp());
return ret;
}
else if (ret == WEBRTC_VIDEO_CODEC_NO_OUTPUT ||
ret == WEBRTC_VIDEO_CODEC_REQUEST_SLI)
{
// No output
_callback->Pop(frame.TimeStamp());
}
// Update the key frame decoded variable so that we know whether or not we've decoded a key frame since reset.
_keyFrameDecoded = (frame.FrameType() == kVideoFrameKey || frame.FrameType() == kVideoFrameGolden);
return ret;
}
WebRtc_Word32
VCMGenericDecoder::Release()
{
_keyFrameDecoded = false;
return _decoder.Release();
}
WebRtc_Word32 VCMGenericDecoder::Reset()
{
_keyFrameDecoded = false;
return _decoder.Reset();
}
WebRtc_Word32 VCMGenericDecoder::SetCodecConfigParameters(const WebRtc_UWord8* buffer, WebRtc_Word32 size)
{
return _decoder.SetCodecConfigParameters(buffer, size);
}
WebRtc_Word32 VCMGenericDecoder::RegisterDecodeCompleteCallback(VCMDecodedFrameCallback* callback)
{
_callback = callback;
return _decoder.RegisterDecodeCompleteCallback(callback);
}
bool VCMGenericDecoder::External() const
{
return _isExternal;
}
} // namespace

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_
#define WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_
#include "timing.h"
#include "timestamp_map.h"
#include "video_codec_interface.h"
#include "encoded_frame.h"
#include "module_common_types.h"
namespace webrtc
{
class VCMReceiveCallback;
enum { kDecoderFrameMemoryLength = 10 };
struct VCMFrameInformation
{
WebRtc_Word64 renderTimeMs;
WebRtc_Word64 decodeStartTimeMs;
void* userData;
};
class VCMDecodedFrameCallback : public DecodedImageCallback
{
public:
VCMDecodedFrameCallback(VCMTiming& timing, TickTimeBase* clock);
virtual ~VCMDecodedFrameCallback();
void SetUserReceiveCallback(VCMReceiveCallback* receiveCallback);
virtual WebRtc_Word32 Decoded(VideoFrame& decodedImage);
virtual WebRtc_Word32 ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId);
virtual WebRtc_Word32 ReceivedDecodedFrame(const WebRtc_UWord64 pictureId);
WebRtc_UWord64 LastReceivedPictureID() const;
WebRtc_Word32 Map(WebRtc_UWord32 timestamp, VCMFrameInformation* frameInfo);
WebRtc_Word32 Pop(WebRtc_UWord32 timestamp);
private:
CriticalSectionWrapper* _critSect;
TickTimeBase* _clock;
VideoFrame _frame;
VCMReceiveCallback* _receiveCallback;
VCMTiming& _timing;
VCMTimestampMap _timestampMap;
WebRtc_UWord64 _lastReceivedPictureID;
};
class VCMGenericDecoder
{
friend class VCMCodecDataBase;
public:
VCMGenericDecoder(VideoDecoder& decoder, WebRtc_Word32 id = 0, bool isExternal = false);
~VCMGenericDecoder();
/**
* Initialize the decoder with the information from the VideoCodec
*/
WebRtc_Word32 InitDecode(const VideoCodec* settings,
WebRtc_Word32 numberOfCores,
bool requireKeyFrame);
/**
* Decode to a raw I420 frame,
*
* inputVideoBuffer reference to encoded video frame
*/
WebRtc_Word32 Decode(const VCMEncodedFrame& inputFrame, int64_t nowMs);
/**
* Free the decoder memory
*/
WebRtc_Word32 Release();
/**
* Reset the decoder state, prepare for a new call
*/
WebRtc_Word32 Reset();
/**
* Codec configuration data sent out-of-band, i.e. in SIP call setup
*
* buffer pointer to the configuration data
* size the size of the configuration data in bytes
*/
WebRtc_Word32 SetCodecConfigParameters(const WebRtc_UWord8* /*buffer*/,
WebRtc_Word32 /*size*/);
WebRtc_Word32 RegisterDecodeCompleteCallback(VCMDecodedFrameCallback* callback);
bool External() const;
protected:
WebRtc_Word32 _id;
VCMDecodedFrameCallback* _callback;
VCMFrameInformation _frameInfos[kDecoderFrameMemoryLength];
WebRtc_UWord32 _nextFrameInfoIdx;
VideoDecoder& _decoder;
VideoCodecType _codecType;
bool _isExternal;
bool _requireKeyFrame;
bool _keyFrameDecoded;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "encoded_frame.h"
#include "generic_encoder.h"
#include "media_optimization.h"
#include "../../../../engine_configurations.h"
namespace webrtc {
//#define DEBUG_ENCODER_BIT_STREAM
VCMGenericEncoder::VCMGenericEncoder(VideoEncoder& encoder, bool internalSource /*= false*/)
:
_encoder(encoder),
_codecType(kVideoCodecUnknown),
_VCMencodedFrameCallback(NULL),
_bitRate(0),
_frameRate(0),
_internalSource(false)
{
}
VCMGenericEncoder::~VCMGenericEncoder()
{
}
WebRtc_Word32 VCMGenericEncoder::Release()
{
_bitRate = 0;
_frameRate = 0;
_VCMencodedFrameCallback = NULL;
return _encoder.Release();
}
WebRtc_Word32
VCMGenericEncoder::InitEncode(const VideoCodec* settings,
WebRtc_Word32 numberOfCores,
WebRtc_UWord32 maxPayloadSize)
{
_bitRate = settings->startBitrate;
_frameRate = settings->maxFramerate;
_codecType = settings->codecType;
if (_VCMencodedFrameCallback != NULL)
{
_VCMencodedFrameCallback->SetCodecType(_codecType);
}
return _encoder.InitEncode(settings, numberOfCores, maxPayloadSize);
}
WebRtc_Word32
VCMGenericEncoder::Encode(const VideoFrame& inputFrame,
const CodecSpecificInfo* codecSpecificInfo,
const std::vector<FrameType>* frameTypes) {
std::vector<VideoFrameType> video_frame_types(frameTypes->size(),
kDeltaFrame);
if (frameTypes) {
VCMEncodedFrame::ConvertFrameTypes(*frameTypes, &video_frame_types);
}
return _encoder.Encode(inputFrame, codecSpecificInfo, &video_frame_types);
}
WebRtc_Word32
VCMGenericEncoder::SetChannelParameters(WebRtc_Word32 packetLoss, int rtt)
{
return _encoder.SetChannelParameters(packetLoss, rtt);
}
WebRtc_Word32
VCMGenericEncoder::SetRates(WebRtc_UWord32 newBitRate, WebRtc_UWord32 frameRate)
{
WebRtc_Word32 ret = _encoder.SetRates(newBitRate, frameRate);
if (ret < 0)
{
return ret;
}
_bitRate = newBitRate;
_frameRate = frameRate;
return VCM_OK;
}
WebRtc_Word32
VCMGenericEncoder::CodecConfigParameters(WebRtc_UWord8* buffer, WebRtc_Word32 size)
{
WebRtc_Word32 ret = _encoder.CodecConfigParameters(buffer, size);
if (ret < 0)
{
return ret;
}
return ret;
}
WebRtc_UWord32 VCMGenericEncoder::BitRate() const
{
return _bitRate;
}
WebRtc_UWord32 VCMGenericEncoder::FrameRate() const
{
return _frameRate;
}
WebRtc_Word32
VCMGenericEncoder::SetPeriodicKeyFrames(bool enable)
{
return _encoder.SetPeriodicKeyFrames(enable);
}
WebRtc_Word32 VCMGenericEncoder::RequestFrame(
const std::vector<FrameType>* frame_types) {
if (!frame_types) {
return 0;
}
VideoFrame image;
std::vector<VideoFrameType> video_frame_types(kVideoFrameDelta);
if (frame_types) {
VCMEncodedFrame::ConvertFrameTypes(*frame_types, &video_frame_types);
}
return _encoder.Encode(image, NULL, &video_frame_types);
}
WebRtc_Word32
VCMGenericEncoder::RegisterEncodeCallback(VCMEncodedFrameCallback* VCMencodedFrameCallback)
{
_VCMencodedFrameCallback = VCMencodedFrameCallback;
_VCMencodedFrameCallback->SetCodecType(_codecType);
_VCMencodedFrameCallback->SetInternalSource(_internalSource);
return _encoder.RegisterEncodeCompleteCallback(_VCMencodedFrameCallback);
}
bool
VCMGenericEncoder::InternalSource() const
{
return _internalSource;
}
/***************************
* Callback Implementation
***************************/
VCMEncodedFrameCallback::VCMEncodedFrameCallback():
_sendCallback(),
_mediaOpt(NULL),
_encodedBytes(0),
_payloadType(0),
_codecType(kVideoCodecUnknown),
_internalSource(false)
#ifdef DEBUG_ENCODER_BIT_STREAM
, _bitStreamAfterEncoder(NULL)
#endif
{
#ifdef DEBUG_ENCODER_BIT_STREAM
_bitStreamAfterEncoder = fopen("encoderBitStream.bit", "wb");
#endif
}
VCMEncodedFrameCallback::~VCMEncodedFrameCallback()
{
#ifdef DEBUG_ENCODER_BIT_STREAM
fclose(_bitStreamAfterEncoder);
#endif
}
WebRtc_Word32
VCMEncodedFrameCallback::SetTransportCallback(VCMPacketizationCallback* transport)
{
_sendCallback = transport;
return VCM_OK;
}
WebRtc_Word32
VCMEncodedFrameCallback::Encoded(
EncodedImage &encodedImage,
const CodecSpecificInfo* codecSpecificInfo,
const RTPFragmentationHeader* fragmentationHeader)
{
FrameType frameType = VCMEncodedFrame::ConvertFrameType(encodedImage._frameType);
WebRtc_UWord32 encodedBytes = 0;
if (_sendCallback != NULL)
{
encodedBytes = encodedImage._length;
#ifdef DEBUG_ENCODER_BIT_STREAM
if (_bitStreamAfterEncoder != NULL)
{
fwrite(encodedImage._buffer, 1, encodedImage._length, _bitStreamAfterEncoder);
}
#endif
RTPVideoHeader rtpVideoHeader;
RTPVideoHeader* rtpVideoHeaderPtr = &rtpVideoHeader;
if (codecSpecificInfo)
{
CopyCodecSpecific(*codecSpecificInfo, &rtpVideoHeaderPtr);
}
else
{
rtpVideoHeaderPtr = NULL;
}
WebRtc_Word32 callbackReturn = _sendCallback->SendData(
frameType,
_payloadType,
encodedImage._timeStamp,
encodedImage.capture_time_ms_,
encodedImage._buffer,
encodedBytes,
*fragmentationHeader,
rtpVideoHeaderPtr);
if (callbackReturn < 0)
{
return callbackReturn;
}
}
else
{
return VCM_UNINITIALIZED;
}
_encodedBytes = encodedBytes;
if (_mediaOpt != NULL) {
_mediaOpt->UpdateWithEncodedData(_encodedBytes, frameType);
if (_internalSource)
{
return _mediaOpt->DropFrame(); // Signal to encoder to drop next frame
}
}
return VCM_OK;
}
WebRtc_UWord32
VCMEncodedFrameCallback::EncodedBytes()
{
return _encodedBytes;
}
void
VCMEncodedFrameCallback::SetMediaOpt(VCMMediaOptimization *mediaOpt)
{
_mediaOpt = mediaOpt;
}
void VCMEncodedFrameCallback::CopyCodecSpecific(const CodecSpecificInfo& info,
RTPVideoHeader** rtp) {
switch (info.codecType) {
case kVideoCodecVP8: {
(*rtp)->codecHeader.VP8.InitRTPVideoHeaderVP8();
(*rtp)->codecHeader.VP8.pictureId =
info.codecSpecific.VP8.pictureId;
(*rtp)->codecHeader.VP8.nonReference =
info.codecSpecific.VP8.nonReference;
(*rtp)->codecHeader.VP8.temporalIdx =
info.codecSpecific.VP8.temporalIdx;
(*rtp)->codecHeader.VP8.layerSync =
info.codecSpecific.VP8.layerSync;
(*rtp)->codecHeader.VP8.tl0PicIdx =
info.codecSpecific.VP8.tl0PicIdx;
(*rtp)->codecHeader.VP8.keyIdx =
info.codecSpecific.VP8.keyIdx;
(*rtp)->simulcastIdx = info.codecSpecific.VP8.simulcastIdx;
return;
}
default: {
// No codec specific info. Change RTP header pointer to NULL.
*rtp = NULL;
return;
}
}
}
} // namespace webrtc

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_GENERIC_ENCODER_H_
#define WEBRTC_MODULES_VIDEO_CODING_GENERIC_ENCODER_H_
#include "video_codec_interface.h"
#include <stdio.h>
namespace webrtc
{
class VCMMediaOptimization;
/*************************************/
/* VCMEncodeFrameCallback class */
/***********************************/
class VCMEncodedFrameCallback : public EncodedImageCallback
{
public:
VCMEncodedFrameCallback();
virtual ~VCMEncodedFrameCallback();
/*
* Callback implementation - codec encode complete
*/
WebRtc_Word32 Encoded(
EncodedImage& encodedImage,
const CodecSpecificInfo* codecSpecificInfo = NULL,
const RTPFragmentationHeader* fragmentationHeader = NULL);
/*
* Get number of encoded bytes
*/
WebRtc_UWord32 EncodedBytes();
/*
* Callback implementation - generic encoder encode complete
*/
WebRtc_Word32 SetTransportCallback(VCMPacketizationCallback* transport);
/**
* Set media Optimization
*/
void SetMediaOpt (VCMMediaOptimization* mediaOpt);
void SetPayloadType(WebRtc_UWord8 payloadType) { _payloadType = payloadType; };
void SetCodecType(VideoCodecType codecType) {_codecType = codecType;};
void SetInternalSource(bool internalSource) { _internalSource = internalSource; };
private:
/*
* Map information from info into rtp. If no relevant information is found
* in info, rtp is set to NULL.
*/
static void CopyCodecSpecific(const CodecSpecificInfo& info,
RTPVideoHeader** rtp);
VCMPacketizationCallback* _sendCallback;
VCMMediaOptimization* _mediaOpt;
WebRtc_UWord32 _encodedBytes;
WebRtc_UWord8 _payloadType;
VideoCodecType _codecType;
bool _internalSource;
#ifdef DEBUG_ENCODER_BIT_STREAM
FILE* _bitStreamAfterEncoder;
#endif
};// end of VCMEncodeFrameCallback class
/******************************/
/* VCMGenericEncoder class */
/******************************/
class VCMGenericEncoder
{
friend class VCMCodecDataBase;
public:
VCMGenericEncoder(VideoEncoder& encoder, bool internalSource = false);
~VCMGenericEncoder();
/**
* Free encoder memory
*/
WebRtc_Word32 Release();
/**
* Initialize the encoder with the information from the VideoCodec
*/
WebRtc_Word32 InitEncode(const VideoCodec* settings,
WebRtc_Word32 numberOfCores,
WebRtc_UWord32 maxPayloadSize);
/**
* Encode raw image
* inputFrame : Frame containing raw image
* codecSpecificInfo : Specific codec data
* cameraFrameRate : request or information from the remote side
* frameType : The requested frame type to encode
*/
WebRtc_Word32 Encode(const VideoFrame& inputFrame,
const CodecSpecificInfo* codecSpecificInfo,
const std::vector<FrameType>* frameTypes);
/**
* Set new target bit rate and frame rate
* Return Value: new bit rate if OK, otherwise <0s
*/
WebRtc_Word32 SetRates(WebRtc_UWord32 newBitRate, WebRtc_UWord32 frameRate);
/**
* Set a new packet loss rate and a new round-trip time in milliseconds.
*/
WebRtc_Word32 SetChannelParameters(WebRtc_Word32 packetLoss, int rtt);
WebRtc_Word32 CodecConfigParameters(WebRtc_UWord8* buffer, WebRtc_Word32 size);
/**
* Register a transport callback which will be called to deliver the encoded buffers
*/
WebRtc_Word32 RegisterEncodeCallback(VCMEncodedFrameCallback* VCMencodedFrameCallback);
/**
* Get encoder bit rate
*/
WebRtc_UWord32 BitRate() const;
/**
* Get encoder frame rate
*/
WebRtc_UWord32 FrameRate() const;
WebRtc_Word32 SetPeriodicKeyFrames(bool enable);
WebRtc_Word32 RequestFrame(const std::vector<FrameType>* frame_types);
bool InternalSource() const;
private:
VideoEncoder& _encoder;
VideoCodecType _codecType;
VCMEncodedFrameCallback* _VCMencodedFrameCallback;
WebRtc_UWord32 _bitRate;
WebRtc_UWord32 _frameRate;
bool _internalSource;
}; // end of VCMGenericEncoder class
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_GENERIC_ENCODER_H_

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "inter_frame_delay.h"
namespace webrtc {
VCMInterFrameDelay::VCMInterFrameDelay(int64_t currentWallClock)
{
Reset(currentWallClock);
}
// Resets the delay estimate
void
VCMInterFrameDelay::Reset(int64_t currentWallClock)
{
_zeroWallClock = currentWallClock;
_wrapArounds = 0;
_prevWallClock = 0;
_prevTimestamp = 0;
_dTS = 0;
}
// Calculates the delay of a frame with the given timestamp.
// This method is called when the frame is complete.
bool
VCMInterFrameDelay::CalculateDelay(WebRtc_UWord32 timestamp,
WebRtc_Word64 *delay,
int64_t currentWallClock)
{
if (_prevWallClock == 0)
{
// First set of data, initialization, wait for next frame
_prevWallClock = currentWallClock;
_prevTimestamp = timestamp;
*delay = 0;
return true;
}
WebRtc_Word32 prevWrapArounds = _wrapArounds;
CheckForWrapArounds(timestamp);
// This will be -1 for backward wrap arounds and +1 for forward wrap arounds
WebRtc_Word32 wrapAroundsSincePrev = _wrapArounds - prevWrapArounds;
// Account for reordering in jitter variance estimate in the future?
// Note that this also captures incomplete frames which are grabbed
// for decoding after a later frame has been complete, i.e. real
// packet losses.
if ((wrapAroundsSincePrev == 0 && timestamp < _prevTimestamp) || wrapAroundsSincePrev < 0)
{
*delay = 0;
return false;
}
// Compute the compensated timestamp difference and convert it to ms and
// round it to closest integer.
_dTS = static_cast<WebRtc_Word64>((timestamp + wrapAroundsSincePrev *
(static_cast<WebRtc_Word64>(1)<<32) - _prevTimestamp) / 90.0 + 0.5);
// frameDelay is the difference of dT and dTS -- i.e. the difference of
// the wall clock time difference and the timestamp difference between
// two following frames.
*delay = static_cast<WebRtc_Word64>(currentWallClock - _prevWallClock - _dTS);
_prevTimestamp = timestamp;
_prevWallClock = currentWallClock;
return true;
}
// Returns the current difference between incoming timestamps
WebRtc_UWord32 VCMInterFrameDelay::CurrentTimeStampDiffMs() const
{
if (_dTS < 0)
{
return 0;
}
return static_cast<WebRtc_UWord32>(_dTS);
}
// Investigates if the timestamp clock has overflowed since the last timestamp and
// keeps track of the number of wrap arounds since reset.
void
VCMInterFrameDelay::CheckForWrapArounds(WebRtc_UWord32 timestamp)
{
if (timestamp < _prevTimestamp)
{
// This difference will probably be less than -2^31 if we have had a wrap around
// (e.g. timestamp = 1, _previousTimestamp = 2^32 - 1). Since it is cast to a Word32,
// it should be positive.
if (static_cast<WebRtc_Word32>(timestamp - _prevTimestamp) > 0)
{
// Forward wrap around
_wrapArounds++;
}
}
// This difference will probably be less than -2^31 if we have had a backward wrap around.
// Since it is cast to a Word32, it should be positive.
else if (static_cast<WebRtc_Word32>(_prevTimestamp - timestamp) > 0)
{
// Backward wrap around
_wrapArounds--;
}
}
}

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_INTER_FRAME_DELAY_H_
#define WEBRTC_MODULES_VIDEO_CODING_INTER_FRAME_DELAY_H_
#include "typedefs.h"
namespace webrtc
{
class VCMInterFrameDelay
{
public:
VCMInterFrameDelay(int64_t currentWallClock);
// Resets the estimate. Zeros are given as parameters.
void Reset(int64_t currentWallClock);
// Calculates the delay of a frame with the given timestamp.
// This method is called when the frame is complete.
//
// Input:
// - timestamp : RTP timestamp of a received frame
// - *delay : Pointer to memory where the result should be stored
// - currentWallClock : The current time in milliseconds.
// Should be -1 for normal operation, only used for testing.
// Return value : true if OK, false when reordered timestamps
bool CalculateDelay(WebRtc_UWord32 timestamp,
WebRtc_Word64 *delay,
int64_t currentWallClock);
// Returns the current difference between incoming timestamps
//
// Return value : Wrap-around compensated difference between incoming
// timestamps.
WebRtc_UWord32 CurrentTimeStampDiffMs() const;
private:
// Controls if the RTP timestamp counter has had a wrap around
// between the current and the previously received frame.
//
// Input:
// - timestmap : RTP timestamp of the current frame.
void CheckForWrapArounds(WebRtc_UWord32 timestamp);
WebRtc_Word64 _zeroWallClock; // Local timestamp of the first video packet received
WebRtc_Word32 _wrapArounds; // Number of wrapArounds detected
// The previous timestamp passed to the delay estimate
WebRtc_UWord32 _prevTimestamp;
// The previous wall clock timestamp used by the delay estimate
WebRtc_Word64 _prevWallClock;
// Wrap-around compensated difference between incoming timestamps
WebRtc_Word64 _dTS;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_INTER_FRAME_DELAY_H_

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_SOURCE_INTERNAL_DEFINES_H_
#define WEBRTC_MODULES_VIDEO_CODING_SOURCE_INTERNAL_DEFINES_H_
#include "typedefs.h"
namespace webrtc
{
#define MASK_32_BITS(x) (0xFFFFFFFF & (x))
inline WebRtc_UWord32 MaskWord64ToUWord32(WebRtc_Word64 w64)
{
return static_cast<WebRtc_UWord32>(MASK_32_BITS(w64));
}
#define VCM_MAX(a, b) (((a) > (b)) ? (a) : (b))
#define VCM_MIN(a, b) (((a) < (b)) ? (a) : (b))
#define VCM_DEFAULT_CODEC_WIDTH 352
#define VCM_DEFAULT_CODEC_HEIGHT 288
#define VCM_DEFAULT_FRAME_RATE 30
#define VCM_MIN_BITRATE 30
#define VCM_FLUSH_INDICATOR 4
// Helper macros for creating the static codec list
#define VCM_NO_CODEC_IDX -1
#ifdef VIDEOCODEC_VP8
#define VCM_VP8_IDX VCM_NO_CODEC_IDX + 1
#else
#define VCM_VP8_IDX VCM_NO_CODEC_IDX
#endif
#ifdef VIDEOCODEC_I420
#define VCM_I420_IDX VCM_VP8_IDX + 1
#else
#define VCM_I420_IDX VCM_VP8_IDX
#endif
#define VCM_NUM_VIDEO_CODECS_AVAILABLE VCM_I420_IDX + 1
#define VCM_NO_RECEIVER_ID 0
inline WebRtc_Word32 VCMId(const WebRtc_Word32 vcmId, const WebRtc_Word32 receiverId = 0)
{
return static_cast<WebRtc_Word32>((vcmId << 16) + receiverId);
}
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_SOURCE_INTERNAL_DEFINES_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_JITTER_BUFFER_H_
#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_JITTER_BUFFER_H_
#include <list>
#include "modules/interface/module_common_types.h"
#include "modules/video_coding/main/interface/video_coding_defines.h"
#include "modules/video_coding/main/source/decoding_state.h"
#include "modules/video_coding/main/source/event.h"
#include "modules/video_coding/main/source/inter_frame_delay.h"
#include "modules/video_coding/main/source/jitter_buffer_common.h"
#include "modules/video_coding/main/source/jitter_estimator.h"
#include "system_wrappers/interface/constructor_magic.h"
#include "system_wrappers/interface/critical_section_wrapper.h"
#include "typedefs.h"
namespace webrtc {
enum VCMNackMode {
kNackInfinite,
kNackHybrid,
kNoNack
};
typedef std::list<VCMFrameBuffer*> FrameList;
// forward declarations
class TickTimeBase;
class VCMFrameBuffer;
class VCMPacket;
class VCMEncodedFrame;
struct VCMJitterSample {
VCMJitterSample() : timestamp(0), frame_size(0), latest_packet_time(-1) {}
uint32_t timestamp;
uint32_t frame_size;
int64_t latest_packet_time;
};
class VCMJitterBuffer {
public:
VCMJitterBuffer(TickTimeBase* clock, int vcm_id = -1, int receiver_id = -1,
bool master = true);
virtual ~VCMJitterBuffer();
// Makes |this| a deep copy of |rhs|.
void CopyFrom(const VCMJitterBuffer& rhs);
// Initializes and starts jitter buffer.
void Start();
// Signals all internal events and stops the jitter buffer.
void Stop();
// Returns true if the jitter buffer is running.
bool Running() const;
// Empty the jitter buffer of all its data.
void Flush();
// Get the number of received key and delta frames since the jitter buffer
// was started.
void FrameStatistics(uint32_t* received_delta_frames,
uint32_t* received_key_frames) const;
// The number of packets discarded by the jitter buffer because the decoder
// won't be able to decode them.
int num_not_decodable_packets() const;
// Gets number of packets discarded by the jitter buffer.
int num_discarded_packets() const;
// Statistics, Calculate frame and bit rates.
void IncomingRateStatistics(unsigned int* framerate,
unsigned int* bitrate);
// Waits for the first packet in the next frame to arrive and then returns
// the timestamp of that frame. |incoming_frame_type| and |render_time_ms| are
// set to the frame type and render time of the next frame.
// Blocks for up to |max_wait_time_ms| ms. Returns -1 if no packet has arrived
// after |max_wait_time_ms| ms.
int64_t NextTimestamp(uint32_t max_wait_time_ms,
FrameType* incoming_frame_type,
int64_t* render_time_ms);
// Checks if the packet sequence will be complete if the next frame would be
// grabbed for decoding. That is, if a frame has been lost between the
// last decoded frame and the next, or if the next frame is missing one
// or more packets.
bool CompleteSequenceWithNextFrame();
// TODO(mikhal/stefan): Merge all GetFrameForDecoding into one.
// Wait |max_wait_time_ms| for a complete frame to arrive. After timeout NULL
// is returned.
VCMEncodedFrame* GetCompleteFrameForDecoding(uint32_t max_wait_time_ms);
// Get a frame for decoding (even an incomplete) without delay.
VCMEncodedFrame* GetFrameForDecoding();
// Releases a frame returned from the jitter buffer, should be called when
// done with decoding.
void ReleaseFrame(VCMEncodedFrame* frame);
// Returns the frame assigned to this timestamp.
int GetFrame(const VCMPacket& packet, VCMEncodedFrame*&);
VCMEncodedFrame* GetFrame(const VCMPacket& packet); // Deprecated.
// Returns the time in ms when the latest packet was inserted into the frame.
// Retransmitted is set to true if any of the packets belonging to the frame
// has been retransmitted.
int64_t LastPacketTime(VCMEncodedFrame* frame, bool* retransmitted) const;
// Inserts a packet into a frame returned from GetFrame().
VCMFrameBufferEnum InsertPacket(VCMEncodedFrame* frame,
const VCMPacket& packet);
// Returns the estimated jitter in milliseconds.
uint32_t EstimatedJitterMs();
// Updates the round-trip time estimate.
void UpdateRtt(uint32_t rtt_ms);
// Set the NACK mode. |highRttNackThreshold| is an RTT threshold in ms above
// which NACK will be disabled if the NACK mode is |kNackHybrid|, -1 meaning
// that NACK is always enabled in the hybrid mode.
// |lowRttNackThreshold| is an RTT threshold in ms below which we expect to
// rely on NACK only, and therefore are using larger buffers to have time to
// wait for retransmissions.
void SetNackMode(VCMNackMode mode, int low_rtt_nack_threshold_ms,
int high_rtt_nack_threshold_ms);
// Returns the current NACK mode.
VCMNackMode nack_mode() const;
// Creates a list of missing sequence numbers.
uint16_t* CreateNackList(uint16_t* nack_list_size, bool* list_extended);
int64_t LastDecodedTimestamp() const;
private:
// In NACK-only mode this function doesn't return or release non-complete
// frames unless we have a complete key frame. In hybrid mode, we may release
// "decodable", incomplete frames.
VCMEncodedFrame* GetFrameForDecodingNACK();
void ReleaseFrameIfNotDecoding(VCMFrameBuffer* frame);
// Gets an empty frame, creating a new frame if necessary (i.e. increases
// jitter buffer size).
VCMFrameBuffer* GetEmptyFrame();
// Recycles oldest frames until a key frame is found. Used if jitter buffer is
// completely full. Returns true if a key frame was found.
bool RecycleFramesUntilKeyFrame();
// Sets the state of |frame| to complete if it's not too old to be decoded.
// Also updates the frame statistics. Signals the |frame_event| if this is
// the next frame to be decoded.
VCMFrameBufferEnum UpdateFrameState(VCMFrameBuffer* frame);
// Finds the oldest complete frame, used for getting next frame to decode.
// Can return a decodable, incomplete frame if |enable_decodable| is true.
FrameList::iterator FindOldestCompleteContinuousFrame(bool enable_decodable);
void CleanUpOldFrames();
// Sets the "decodable" and "frame loss" flags of a frame depending on which
// packets have been received and which are missing.
// A frame is "decodable" if enough packets of that frame has been received
// for it to be usable by the decoder.
// A frame has the "frame loss" flag set if packets are missing after the
// last decoded frame and before |frame|.
void VerifyAndSetPreviousFrameLost(VCMFrameBuffer* frame);
// Returns true if |packet| is likely to have been retransmitted.
bool IsPacketRetransmitted(const VCMPacket& packet) const;
// The following three functions update the jitter estimate with the
// payload size, receive time and RTP timestamp of a frame.
void UpdateJitterEstimate(const VCMJitterSample& sample,
bool incomplete_frame);
void UpdateJitterEstimate(const VCMFrameBuffer& frame, bool incomplete_frame);
void UpdateJitterEstimate(int64_t latest_packet_time_ms,
uint32_t timestamp,
unsigned int frame_size,
bool incomplete_frame);
// Returns the lowest and highest known sequence numbers, where the lowest is
// the last decoded sequence number if a frame has been decoded.
// -1 is returned if a sequence number cannot be determined.
void GetLowHighSequenceNumbers(int32_t* low_seq_num,
int32_t* high_seq_num) const;
// Returns true if we should wait for retransmissions, false otherwise.
bool WaitForRetransmissions();
int vcm_id_;
int receiver_id_;
TickTimeBase* clock_;
// If we are running (have started) or not.
bool running_;
CriticalSectionWrapper* crit_sect_;
bool master_;
// Event to signal when we have a frame ready for decoder.
VCMEvent frame_event_;
// Event to signal when we have received a packet.
VCMEvent packet_event_;
// Number of allocated frames.
int max_number_of_frames_;
// Array of pointers to the frames in jitter buffer.
VCMFrameBuffer* frame_buffers_[kMaxNumberOfFrames];
FrameList frame_list_;
VCMDecodingState last_decoded_state_;
bool first_packet_;
// Statistics.
int num_not_decodable_packets_;
// Frame counter for each type (key, delta, golden, key-delta).
unsigned int receive_statistics_[4];
// Latest calculated frame rates of incoming stream.
unsigned int incoming_frame_rate_;
unsigned int incoming_frame_count_;
int64_t time_last_incoming_frame_count_;
unsigned int incoming_bit_count_;
unsigned int incoming_bit_rate_;
unsigned int drop_count_; // Frame drop counter.
// Number of frames in a row that have been too old.
int num_consecutive_old_frames_;
// Number of packets in a row that have been too old.
int num_consecutive_old_packets_;
// Number of packets discarded by the jitter buffer.
int num_discarded_packets_;
// Jitter estimation.
// Filter for estimating jitter.
VCMJitterEstimator jitter_estimate_;
// Calculates network delays used for jitter calculations.
VCMInterFrameDelay inter_frame_delay_;
VCMJitterSample waiting_for_completion_;
WebRtc_UWord32 rtt_ms_;
// NACK and retransmissions.
VCMNackMode nack_mode_;
int low_rtt_nack_threshold_ms_;
int high_rtt_nack_threshold_ms_;
// Holds the internal NACK list (the missing sequence numbers).
int32_t nack_seq_nums_internal_[kNackHistoryLength];
uint16_t nack_seq_nums_[kNackHistoryLength];
unsigned int nack_seq_nums_length_;
bool waiting_for_key_frame_;
DISALLOW_COPY_AND_ASSIGN(VCMJitterBuffer);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_JITTER_BUFFER_H_

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