generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

Update talk to 58174641 together with http://review.webrtc.org/4319005/.

Browse Source 
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5287 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
wu@webrtc.org 2013-12-13 19:17:43 +00:00
parent 92c2793154
commit 24301a67c6
28 changed files with 697 additions and 48 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

18
talk/media/webrtc/fakewebrtcvideoengine.h
View File

@ -339,12 +339,14 @@ class FakeWebRtcVideoEngine
};
class Capturer : public webrtc::ViEExternalCapture {
public:
Capturer() : channel_id_(-1), denoising_(false), last_capture_time_(0) { }
Capturer() : channel_id_(-1), denoising_(false),
last_capture_time_(0), incoming_frame_num_(0) { }
int channel_id() const { return channel_id_; }
void set_channel_id(int channel_id) { channel_id_ = channel_id; }
bool denoising() const { return denoising_; }
void set_denoising(bool denoising) { denoising_ = denoising; }
int64 last_capture_time() { return last_capture_time_; }
int64 last_capture_time() const { return last_capture_time_; }
int incoming_frame_num() const { return incoming_frame_num_; }
// From ViEExternalCapture
virtual int IncomingFrame(unsigned char* videoFrame,
@ -359,6 +361,7 @@ class FakeWebRtcVideoEngine
const webrtc::ViEVideoFrameI420& video_frame,
unsigned long long captureTime) {
last_capture_time_ = captureTime;
++incoming_frame_num_;
return 0;
}
@ -366,6 +369,7 @@ class FakeWebRtcVideoEngine
int channel_id_;
bool denoising_;
int64 last_capture_time_;
int incoming_frame_num_;
};
FakeWebRtcVideoEngine(const cricket::VideoCodec* const* codecs,
@ -408,6 +412,16 @@ class FakeWebRtcVideoEngine
int GetLastCapturer() const { return last_capturer_; }
int GetNumCapturers() const { return static_cast<int>(capturers_.size()); }
int GetIncomingFrameNum(int channel_id) const {
for (std::map<int, Capturer*>::const_iterator iter = capturers_.begin();
iter != capturers_.end(); ++iter) {
Capturer* capturer = iter->second;
if (capturer->channel_id() == channel_id) {
return capturer->incoming_frame_num();
}
}
return -1;
}
void set_fail_alloc_capturer(bool fail_alloc_capturer) {
fail_alloc_capturer_ = fail_alloc_capturer;
}

7
talk/media/webrtc/fakewebrtcvoiceengine.h
View File

@ -631,6 +631,13 @@ class FakeWebRtcVoiceEngine
// webrtc::VoENetEqStats
WEBRTC_STUB(GetNetworkStatistics, (int, webrtc::NetworkStatistics&));
#ifdef USE_WEBRTC_DEV_BRANCH
WEBRTC_FUNC_CONST(GetDecodingCallStatistics, (int channel,
webrtc::AudioDecodingCallStats*)) {
WEBRTC_CHECK_CHANNEL(channel);
return 0;
}
#endif
// webrtc::VoENetwork
WEBRTC_FUNC(RegisterExternalTransport, (int channel,

3
talk/media/webrtc/webrtcmediaengine.h
View File

@ -145,6 +145,9 @@ class WebRtcMediaEngine : public cricket::MediaEngineInterface {
virtual void SetVideoLogging(int min_sev, const char* filter) OVERRIDE {
delegate_->SetVideoLogging(min_sev, filter);
}
virtual bool StartAecDump(FILE* file) OVERRIDE {
return delegate_->StartAecDump(file);
}
virtual bool RegisterVoiceProcessor(
uint32 ssrc, VoiceProcessor* video_processor,
MediaProcessorDirection direction) OVERRIDE {

77
talk/media/webrtc/webrtcvideoengine.cc
View File

@ -2118,18 +2118,6 @@ bool WebRtcVideoMediaChannel::GetSendChannelKey(uint32 local_ssrc,
return true;
}
WebRtcVideoChannelSendInfo* WebRtcVideoMediaChannel::GetSendChannel(
VideoCapturer* video_capturer) {
for (SendChannelMap::iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
WebRtcVideoChannelSendInfo* send_channel = iter->second;
if (send_channel->video_capturer() == video_capturer) {
return send_channel;
}
}
return NULL;
}
WebRtcVideoChannelSendInfo* WebRtcVideoMediaChannel::GetSendChannel(
uint32 local_ssrc) {
uint32 key;
@ -2159,6 +2147,18 @@ bool WebRtcVideoMediaChannel::CreateSendChannelKey(uint32 local_ssrc,
return true;
}
int WebRtcVideoMediaChannel::GetSendChannelNum(VideoCapturer* capturer) {
int num = 0;
for (SendChannelMap::iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
WebRtcVideoChannelSendInfo* send_channel = iter->second;
if (send_channel->video_capturer() == capturer) {
++num;
}
}
return num;
}
uint32 WebRtcVideoMediaChannel::GetDefaultChannelSsrc() {
WebRtcVideoChannelSendInfo* send_channel = send_channels_[0];
const StreamParams* sp = send_channel->stream_params();
@ -2174,11 +2174,8 @@ bool WebRtcVideoMediaChannel::DeleteSendChannel(uint32 ssrc_key) {
return false;
}
WebRtcVideoChannelSendInfo* send_channel = send_channels_[ssrc_key];
VideoCapturer* capturer = send_channel->video_capturer();
if (capturer != NULL) {
capturer->SignalVideoFrame.disconnect(this);
send_channel->set_video_capturer(NULL);
}
MaybeDisconnectCapturer(send_channel->video_capturer());
send_channel->set_video_capturer(NULL);
int channel_id = send_channel->channel_id();
int capture_id = send_channel->capture_id();
@ -2217,7 +2214,7 @@ bool WebRtcVideoMediaChannel::RemoveCapturer(uint32 ssrc) {
if (capturer == NULL) {
return false;
}
capturer->SignalVideoFrame.disconnect(this);
MaybeDisconnectCapturer(capturer);
send_channel->set_video_capturer(NULL);
const int64 timestamp = send_channel->local_stream_info()->time_stamp();
if (send_codec_) {
@ -2468,14 +2465,10 @@ bool WebRtcVideoMediaChannel::SetCapturer(uint32 ssrc,
return false;
}
VideoCapturer* old_capturer = send_channel->video_capturer();
if (old_capturer) {
old_capturer->SignalVideoFrame.disconnect(this);
}
MaybeDisconnectCapturer(old_capturer);
send_channel->set_video_capturer(capturer);
capturer->SignalVideoFrame.connect(
this,
&WebRtcVideoMediaChannel::SendFrame);
MaybeConnectCapturer(capturer);
if (!capturer->IsScreencast() && ratio_w_ != 0 && ratio_h_ != 0) {
capturer->UpdateAspectRatio(ratio_w_, ratio_h_);
}
@ -2865,20 +2858,23 @@ bool WebRtcVideoMediaChannel::GetRenderer(uint32 ssrc,
return true;
}
// TODO(zhurunz): Add unittests to test this function.
// TODO(thorcarpenter): This is broken. One capturer registered on two ssrc
// will not send any video to the second ssrc send channel. We should remove
// GetSendChannel(capturer) and pass in an ssrc here.
void WebRtcVideoMediaChannel::SendFrame(VideoCapturer* capturer,
const VideoFrame* frame) {
// If there's send channel registers to the |capturer|, then only send the
// frame to that channel and return. Otherwise send the frame to the default
// channel, which currently taking frames from the engine.
WebRtcVideoChannelSendInfo* send_channel = GetSendChannel(capturer);
if (send_channel) {
SendFrame(send_channel, frame, capturer->IsScreencast());
// If the |capturer| is registered to any send channel, then send the frame
// to those send channels.
bool capturer_is_channel_owned = false;
for (SendChannelMap::iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
WebRtcVideoChannelSendInfo* send_channel = iter->second;
if (send_channel->video_capturer() == capturer) {
SendFrame(send_channel, frame, capturer->IsScreencast());
capturer_is_channel_owned = true;
}
}
if (capturer_is_channel_owned) {
return;
}
// TODO(hellner): Remove below for loop once the captured frame no longer
// come from the engine, i.e. the engine no longer owns a capturer.
for (SendChannelMap::iterator iter = send_channels_.begin();
@ -3754,6 +3750,19 @@ bool WebRtcVideoMediaChannel::SetLocalRtxSsrc(int channel_id,
return true;
}
void WebRtcVideoMediaChannel::MaybeConnectCapturer(VideoCapturer* capturer) {
if (capturer != NULL && GetSendChannelNum(capturer) == 1) {
capturer->SignalVideoFrame.connect(this,
&WebRtcVideoMediaChannel::SendFrame);
}
}
void WebRtcVideoMediaChannel::MaybeDisconnectCapturer(VideoCapturer* capturer) {
if (capturer != NULL && GetSendChannelNum(capturer) == 1) {
capturer->SignalVideoFrame.disconnect(this);
}
}
} // namespace cricket
#endif // HAVE_WEBRTC_VIDEO

10
talk/media/webrtc/webrtcvideoengine.h
View File

@ -366,11 +366,12 @@ class WebRtcVideoMediaChannel : public talk_base::MessageHandler,
// If the local ssrc correspond to that of the default channel the key is 0.
// For all other channels the returned key will be the same as the local ssrc.
bool GetSendChannelKey(uint32 local_ssrc, uint32* key);
WebRtcVideoChannelSendInfo* GetSendChannel(VideoCapturer* video_capturer);
WebRtcVideoChannelSendInfo* GetSendChannel(uint32 local_ssrc);
// Creates a new unique key that can be used for inserting a new send channel
// into |send_channels_|
bool CreateSendChannelKey(uint32 local_ssrc, uint32* key);
// Get the number of the send channels |capturer| registered with.
int GetSendChannelNum(VideoCapturer* capturer);
bool IsDefaultChannel(int channel_id) const {
return channel_id == vie_channel_;
@ -404,6 +405,13 @@ class WebRtcVideoMediaChannel : public talk_base::MessageHandler,
bool SetLocalRtxSsrc(int channel_id, const StreamParams& send_params,
uint32 primary_ssrc, int stream_idx);
// Connect |capturer| to WebRtcVideoMediaChannel if it is only registered
// to one send channel, i.e. the first send channel.
void MaybeConnectCapturer(VideoCapturer* capturer);
// Disconnect |capturer| from WebRtcVideoMediaChannel if it is only registered
// to one send channel, i.e. the last send channel.
void MaybeDisconnectCapturer(VideoCapturer* capturer);
// Global state.
WebRtcVideoEngine* engine_;
VoiceMediaChannel* voice_channel_;

47
talk/media/webrtc/webrtcvideoengine_unittest.cc
View File

@ -1216,6 +1216,53 @@ TEST_F(WebRtcVideoEngineTestFake, SetOptionsWithDenoising) {
EXPECT_FALSE(vie_.GetCaptureDenoising(capture_id));
}
TEST_F(WebRtcVideoEngineTestFake, MultipleSendStreamsWithOneCapturer) {
EXPECT_TRUE(SetupEngine());
// Start the capturer
cricket::FakeVideoCapturer capturer;
cricket::VideoFormat capture_format_vga = cricket::VideoFormat(640, 480,
cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420);
EXPECT_EQ(cricket::CS_RUNNING, capturer.Start(capture_format_vga));
// Add send streams and connect the capturer
for (unsigned int i = 0; i < sizeof(kSsrcs2)/sizeof(kSsrcs2[0]); ++i) {
EXPECT_TRUE(channel_->AddSendStream(
cricket::StreamParams::CreateLegacy(kSsrcs2[i])));
// Register the capturer to the ssrc.
EXPECT_TRUE(channel_->SetCapturer(kSsrcs2[i], &capturer));
}
const int channel0 = vie_.GetChannelFromLocalSsrc(kSsrcs2[0]);
ASSERT_NE(-1, channel0);
const int channel1 = vie_.GetChannelFromLocalSsrc(kSsrcs2[1]);
ASSERT_NE(-1, channel1);
ASSERT_NE(channel0, channel1);
// Set send codec.
std::vector<cricket::VideoCodec> codecs;
cricket::VideoCodec send_codec(100, "VP8", 640, 480, 30, 0);
codecs.push_back(send_codec);
EXPECT_TRUE(channel_->SetSendCodecs(codecs));
EXPECT_TRUE(capturer.CaptureFrame());
EXPECT_EQ(1, vie_.GetIncomingFrameNum(channel0));
EXPECT_EQ(1, vie_.GetIncomingFrameNum(channel1));
EXPECT_TRUE(channel_->RemoveSendStream(kSsrcs2[0]));
EXPECT_TRUE(capturer.CaptureFrame());
// channel0 is the default channel, so it won't be deleted.
// But it should be disconnected from the capturer.
EXPECT_EQ(1, vie_.GetIncomingFrameNum(channel0));
EXPECT_EQ(2, vie_.GetIncomingFrameNum(channel1));
EXPECT_TRUE(channel_->RemoveSendStream(kSsrcs2[1]));
EXPECT_TRUE(capturer.CaptureFrame());
EXPECT_EQ(1, vie_.GetIncomingFrameNum(channel0));
// channel1 has already been deleted.
EXPECT_EQ(-1, vie_.GetIncomingFrameNum(channel1));
}
// Disabled since its flaky: b/11288120
TEST_F(WebRtcVideoEngineTestFake, DISABLED_SendReceiveBitratesStats) {

19
webrtc/common_types.h
View File

@ -382,6 +382,25 @@ struct NetworkStatistics // NETEQ statistics
int addedSamples;
};
// Statistics for calls to AudioCodingModule::PlayoutData10Ms().
struct AudioDecodingCallStats {
AudioDecodingCallStats()
: calls_to_silence_generator(0),
calls_to_neteq(0),
decoded_normal(0),
decoded_plc(0),
decoded_cng(0),
decoded_plc_cng(0) {}
int calls_to_silence_generator; // Number of calls where silence generated,
// and NetEq was disengaged from decoding.
int calls_to_neteq; // Number of calls to NetEq.
int decoded_normal; // Number of calls where audio RTP packet decoded.
int decoded_plc; // Number of calls resulted in PLC.
int decoded_cng; // Number of calls where comfort noise generated due to DTX.
int decoded_plc_cng; // Number of calls resulted where PLC faded to CNG.
};
typedef struct
{
int min; // minumum

11
webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
View File

@ -19,6 +19,7 @@
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
#include "webrtc/modules/audio_coding/main/acm2/nack.h"
#include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h"
#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h"
@ -461,6 +462,7 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
audio_frame->vad_activity_ = previous_audio_activity_;
SetAudioFrameActivityAndType(vad_enabled_, type, audio_frame);
previous_audio_activity_ = audio_frame->vad_activity_;
call_stats_.DecodedByNetEq(audio_frame->speech_type_);
return 0;
}
@ -761,6 +763,9 @@ bool AcmReceiver::GetSilence(int desired_sample_rate_hz, AudioFrame* frame) {
return false;
}
// Update statistics.
call_stats_.DecodedBySilenceGenerator();
// Set the values if already got a packet, otherwise set to default values.
if (last_audio_decoder_ >= 0) {
current_sample_rate_hz_ = ACMCodecDB::database_[last_audio_decoder_].plfreq;
@ -832,6 +837,12 @@ void AcmReceiver::InsertStreamOfSyncPackets(
}
}
void AcmReceiver::GetDecodingCallStatistics(
AudioDecodingCallStats* stats) const {
CriticalSectionScoped lock(neteq_crit_sect_);
*stats = call_stats_.GetDecodingStatistics();
}
} // namespace acm2
} // namespace webrtc

7
webrtc/modules/audio_coding/main/acm2/acm_receiver.h
View File

@ -18,6 +18,7 @@
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
#include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h"
#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h"
#include "webrtc/modules/interface/module_common_types.h"
@ -320,6 +321,10 @@ class AcmReceiver {
//
NetEqBackgroundNoiseMode BackgroundNoiseModeForTest() const;
//
// Get statistics of calls to GetAudio().
void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
private:
int PayloadType2CodecIndex(uint8_t payload_type) const;
@ -361,6 +366,8 @@ class AcmReceiver {
// initial delay is set.
scoped_ptr<InitialDelayManager::SyncStream> missing_packets_sync_stream_;
scoped_ptr<InitialDelayManager::SyncStream> late_packets_sync_stream_;
CallStatistics call_stats_;
};
} // namespace acm2

2
webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi
View File

@ -84,6 +84,8 @@
'audio_coding_module.cc',
'audio_coding_module_impl.cc',
'audio_coding_module_impl.h',
'call_statistics.cc',
'call_statistics.h',
'initial_delay_manager.cc',
'initial_delay_manager.h',
'nack.cc',

6
webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
View File

@ -20,6 +20,7 @@
#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h"
@ -1979,6 +1980,11 @@ const char* AudioCodingModuleImpl::Version() const {
return kExperimentalAcmVersion;
}
void AudioCodingModuleImpl::GetDecodingCallStatistics(
AudioDecodingCallStats* call_stats) const {
receiver_.GetDecodingCallStatistics(call_stats);
}
} // namespace acm2
} // namespace webrtc

2
webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
View File

@ -228,6 +228,8 @@ class AudioCodingModuleImpl : public AudioCodingModule {
std::vector<uint16_t> GetNackList(int round_trip_time_ms) const;
void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
private:
int UnregisterReceiveCodecSafe(int payload_type);

55
webrtc/modules/audio_coding/main/acm2/call_statistics.cc Normal file
View File

@ -0,0 +1,55 @@
/*
* Copyright (c) 2013 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 "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
#include <cassert>
namespace webrtc {
namespace acm2 {
void CallStatistics::DecodedByNetEq(AudioFrame::SpeechType speech_type) {
++decoding_stat_.calls_to_neteq;
switch (speech_type) {
case AudioFrame::kNormalSpeech: {
++decoding_stat_.decoded_normal;
break;
}
case AudioFrame::kPLC: {
++decoding_stat_.decoded_plc;
break;
}
case AudioFrame::kCNG: {
++decoding_stat_.decoded_cng;
break;
}
case AudioFrame::kPLCCNG: {
++decoding_stat_.decoded_plc_cng;
break;
}
case AudioFrame::kUndefined: {
// If the audio is decoded by NetEq, |kUndefined| is not an option.
assert(false);
}
}
}
void CallStatistics::DecodedBySilenceGenerator() {
++decoding_stat_.calls_to_silence_generator;
}
const AudioDecodingCallStats& CallStatistics::GetDecodingStatistics() const {
return decoding_stat_;
}
} // namespace acm2
} // namespace webrtc

63
webrtc/modules/audio_coding/main/acm2/call_statistics.h Normal file
View File

@ -0,0 +1,63 @@
/*
* Copyright (c) 2013 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_AUDIO_CODING_MAIN_ACM2_CALL_STATISTICS_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CALL_STATISTICS_H_
#include "webrtc/common_types.h"
#include "webrtc/modules/interface/module_common_types.h"
//
// This class is for book keeping of calls to ACM. It is not useful to log API
// calls which are supposed to be called every 10ms, e.g. PlayoutData10Ms(),
// however, it is useful to know the number of such calls in a given time
// interval. The current implementation covers calls to PlayoutData10Ms() with
// detailed accounting of the decoded speech type.
//
// Thread Safety
// =============
// Please note that this class in not thread safe. The class must be protected
// if different APIs are called from different threads.
//
namespace webrtc {
namespace acm2 {
class CallStatistics {
public:
CallStatistics() {}
~CallStatistics() {}
// Call this method to indicate that NetEq engaged in decoding. |speech_type|
// is the audio-type according to NetEq.
void DecodedByNetEq(AudioFrame::SpeechType speech_type);
// Call this method to indicate that a decoding call resulted in generating
// silence, i.e. call to NetEq is bypassed and the output audio is zero.
void DecodedBySilenceGenerator();
// Get statistics for decoding. The statistics include the number of calls to
// NetEq and silence generator, as well as the type of speech pulled of off
// NetEq, c.f. declaration of AudioDecodingCallStats for detailed description.
const AudioDecodingCallStats& GetDecodingStatistics() const;
private:
// Reset the decoding statistics.
void ResetDecodingStatistics();
AudioDecodingCallStats decoding_stat_;
};
} // namespace acm2
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CALL_STATISTICS_H_

55
webrtc/modules/audio_coding/main/acm2/call_statistics_unittest.cc Normal file
View File

@ -0,0 +1,55 @@
/*
* Copyright (c) 2013 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 "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
namespace webrtc {
namespace acm2 {
TEST(CallStatisticsTest, InitializedZero) {
CallStatistics call_stats;
AudioDecodingCallStats stats;
stats = call_stats.GetDecodingStatistics();
EXPECT_EQ(0, stats.calls_to_neteq);
EXPECT_EQ(0, stats.calls_to_silence_generator);
EXPECT_EQ(0, stats.decoded_normal);
EXPECT_EQ(0, stats.decoded_cng);
EXPECT_EQ(0, stats.decoded_plc);
EXPECT_EQ(0, stats.decoded_plc_cng);
}
TEST(CallStatisticsTest, AllCalls) {
CallStatistics call_stats;
AudioDecodingCallStats stats;
call_stats.DecodedBySilenceGenerator();
call_stats.DecodedByNetEq(AudioFrame::kNormalSpeech);
call_stats.DecodedByNetEq(AudioFrame::kPLC);
call_stats.DecodedByNetEq(AudioFrame::kPLCCNG);
call_stats.DecodedByNetEq(AudioFrame::kCNG);
stats = call_stats.GetDecodingStatistics();
EXPECT_EQ(4, stats.calls_to_neteq);
EXPECT_EQ(1, stats.calls_to_silence_generator);
EXPECT_EQ(1, stats.decoded_normal);
EXPECT_EQ(1, stats.decoded_cng);
EXPECT_EQ(1, stats.decoded_plc);
EXPECT_EQ(1, stats.decoded_plc_cng);
}
} // namespace acm2
} // namespace webrtc

3
webrtc/modules/audio_coding/main/interface/audio_coding_module.h
View File

@ -931,6 +931,9 @@ class AudioCodingModule: public Module {
// is returned.
//
virtual std::vector<uint16_t> GetNackList(int round_trip_time_ms) const = 0;
virtual void GetDecodingCallStatistics(
AudioDecodingCallStats* call_stats) const = 0;
};
struct AudioCodingModuleFactory {

13
webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
View File

@ -18,6 +18,7 @@
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
#include "webrtc/modules/audio_coding/main/source/acm_dtmf_detection.h"
#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
#include "webrtc/modules/audio_coding/main/source/acm_resampler.h"
@ -2273,6 +2274,9 @@ int32_t AudioCodingModuleImpl::PlayoutData10Ms(
{
CriticalSectionScoped lock(acm_crit_sect_);
// Update call statistics.
call_stats_.DecodedByNetEq(audio_frame->speech_type_);
if (update_nack) {
assert(nack_.get());
nack_->UpdateLastDecodedPacket(decoded_seq_num, decoded_timestamp);
@ -2879,6 +2883,9 @@ bool AudioCodingModuleImpl::GetSilence(int desired_sample_rate_hz,
return false;
}
// Record call to silence generator.
call_stats_.DecodedBySilenceGenerator();
// We stop accumulating packets, if the number of packets or the total size
// exceeds a threshold.
int max_num_packets;
@ -3030,6 +3037,12 @@ const char* AudioCodingModuleImpl::Version() const {
return kLegacyAcmVersion;
}
void AudioCodingModuleImpl::GetDecodingCallStatistics(
AudioDecodingCallStats* call_stats) const {
CriticalSectionScoped lock(acm_crit_sect_);
*call_stats = call_stats_.GetDecodingStatistics();
}
} // namespace acm1
} // namespace webrtc

5
webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h
View File

@ -19,6 +19,7 @@
#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
#include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
#include "webrtc/modules/audio_coding/main/source/acm_resampler.h"
#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
@ -303,6 +304,8 @@ class AudioCodingModuleImpl : public AudioCodingModule {
// Disable NACK.
void DisableNack();
void GetDecodingCallStatistics(AudioDecodingCallStats* call_stats) const;
private:
// Change required states after starting to receive the codec corresponding
// to |index|.
@ -441,6 +444,8 @@ class AudioCodingModuleImpl : public AudioCodingModule {
Clock* clock_;
scoped_ptr<acm2::Nack> nack_;
bool nack_enabled_;
acm2::CallStatistics call_stats_;
};
} // namespace acm1

16
webrtc/modules/audio_coding/neteq/webrtc_neteq.c
View File

@ -1104,14 +1104,6 @@ int WebRtcNetEQ_GetSpeechOutputType(void *inst, enum WebRtcNetEQOutputType *outp
/* If CN or internal CNG */
*outputType = kOutputCNG;
#ifdef NETEQ_VAD
}
else if ( NetEqMainInst->DSPinst.VADInst.VADDecision == 0 )
{
/* post-decode VAD says passive speaker */
*outputType = kOutputVADPassive;
#endif /* NETEQ_VAD */
}
else if ((NetEqMainInst->DSPinst.w16_mode == MODE_EXPAND)
&& (NetEqMainInst->DSPinst.ExpandInst.w16_expandMuteFactor == 0))
@ -1125,6 +1117,14 @@ int WebRtcNetEQ_GetSpeechOutputType(void *inst, enum WebRtcNetEQOutputType *outp
/* PLC mode */
*outputType = kOutputPLC;
#ifdef NETEQ_VAD
}
else if ( NetEqMainInst->DSPinst.VADInst.VADDecision == 0 )
{
/* post-decode VAD says passive speaker */
*outputType = kOutputVADPassive;
#endif /* NETEQ_VAD */
}
else
{

4
webrtc/modules/audio_coding/neteq4/neteq_impl.cc
View File

@ -1887,13 +1887,13 @@ NetEqOutputType NetEqImpl::LastOutputType() {
assert(expand_.get());
if (last_mode_ == kModeCodecInternalCng || last_mode_ == kModeRfc3389Cng) {
return kOutputCNG;
} else if (vad_->running() && !vad_->active_speech()) {
return kOutputVADPassive;
} else if (last_mode_ == kModeExpand && expand_->MuteFactor(0) == 0) {
// Expand mode has faded down to background noise only (very long expand).
return kOutputPLCtoCNG;
} else if (last_mode_ == kModeExpand) {
return kOutputPLC;
} else if (vad_->running() && !vad_->active_speech()) {
return kOutputVADPassive;
} else {
return kOutputNormal;
}

1
webrtc/modules/modules.gyp
View File

@ -102,6 +102,7 @@
],
'sources': [
'audio_coding/main/acm2/acm_receiver_unittest.cc',
'audio_coding/main/acm2/call_statistics_unittest.cc',
'audio_coding/main/acm2/initial_delay_manager_unittest.cc',
'audio_coding/main/acm2/nack_unittest.cc',
'audio_coding/main/source/acm_neteq_unittest.cc',

4
webrtc/voice_engine/channel.cc
View File

@ -4619,6 +4619,10 @@ Channel::GetNetworkStatistics(NetworkStatistics& stats)
return return_value;
}
void Channel::GetDecodingCallStatistics(AudioDecodingCallStats* stats) const {
audio_coding_->GetDecodingCallStatistics(stats);
}
bool Channel::GetDelayEstimate(int* jitter_buffer_delay_ms,
int* playout_buffer_delay_ms) const {
if (_average_jitter_buffer_delay_us == 0) {

1
webrtc/voice_engine/channel.h
View File

@ -201,6 +201,7 @@ public:
// VoENetEqStats
int GetNetworkStatistics(NetworkStatistics& stats);
void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
// VoEVideoSync
bool GetDelayEstimate(int* jitter_buffer_delay_ms,

4
webrtc/voice_engine/include/voe_neteq_stats.h
View File

@ -35,6 +35,10 @@ public:
// The statistics are reset after the query.
virtual int GetNetworkStatistics(int channel, NetworkStatistics& stats) = 0;
// Get statistics of calls to AudioCodingModule::PlayoutData10Ms().
virtual int GetDecodingCallStatistics(
int channel, AudioDecodingCallStats* stats) const = 0;
protected:
VoENetEqStats() {}
virtual ~VoENetEqStats() {}

22
webrtc/voice_engine/voe_neteq_stats_impl.cc
View File

@ -17,7 +17,6 @@
#include "webrtc/voice_engine/include/voe_errors.h"
#include "webrtc/voice_engine/voice_engine_impl.h"
namespace webrtc {
VoENetEqStats* VoENetEqStats::GetInterface(VoiceEngine* voiceEngine)
@ -73,6 +72,27 @@ int VoENetEqStatsImpl::GetNetworkStatistics(int channel,
return channelPtr->GetNetworkStatistics(stats);
}
int VoENetEqStatsImpl::GetDecodingCallStatistics(
int channel, AudioDecodingCallStats* stats) const {
ANDROID_NOT_SUPPORTED(_shared->statistics());
if (!_shared->statistics().Initialized()) {
_shared->SetLastError(VE_NOT_INITED, kTraceError);
return -1;
}
voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
voe::Channel* channelPtr = ch.channel();
if (channelPtr == NULL) {
_shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
"GetDecodingCallStatistics() failed to locate "
"channel");
return -1;
}
channelPtr->GetDecodingCallStatistics(stats);
return 0;
}
#endif // #ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
} // namespace webrtc

4
webrtc/voice_engine/voe_neteq_stats_impl.h
View File

@ -13,6 +13,7 @@
#include "webrtc/voice_engine/include/voe_neteq_stats.h"
#include "webrtc/common_types.h"
#include "webrtc/voice_engine/shared_data.h"
namespace webrtc {
@ -23,6 +24,9 @@ public:
virtual int GetNetworkStatistics(int channel,
NetworkStatistics& stats);
virtual int GetDecodingCallStatistics(
int channel, AudioDecodingCallStats* stats) const;
protected:
VoENetEqStatsImpl(voe::SharedData* shared);
virtual ~VoENetEqStatsImpl();

285
webrtc/voice_engine/voe_neteq_stats_unittest.cc Normal file
View File

@ -0,0 +1,285 @@
/*
* Copyright (c) 2013 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 "webrtc/voice_engine/include/voe_neteq_stats.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_device/include/fake_audio_device.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/test/testsupport/gtest_disable.h"
#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/voice_engine/include/voe_hardware.h"
#include "webrtc/voice_engine/voice_engine_defines.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h"
#include "webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h"
namespace webrtc {
namespace voe {
namespace {
const int kSampleRateHz = 16000;
const int kNumSamples10ms = kSampleRateHz / 100;
const int kFrameSizeMs = 10; // Multiple of 10.
const int kFrameSizeSamples = kFrameSizeMs / 10 * kNumSamples10ms;
const int kPayloadSizeBytes = kFrameSizeSamples * sizeof(int16_t);
const uint8_t kPayloadType = 111;
class RtpUtility {
public:
RtpUtility(int samples_per_packet, uint8_t payload_type)
: samples_per_packet_(samples_per_packet), payload_type_(payload_type) {}
virtual ~RtpUtility() {}
void Populate(WebRtcRTPHeader* rtp_header) {
rtp_header->header.sequenceNumber = 0xABCD;
rtp_header->header.timestamp = 0xABCDEF01;
rtp_header->header.payloadType = payload_type_;
rtp_header->header.markerBit = false;
rtp_header->header.ssrc = 0x1234;
rtp_header->header.numCSRCs = 0;
rtp_header->frameType = kAudioFrameSpeech;
rtp_header->header.payload_type_frequency = kSampleRateHz;
rtp_header->type.Audio.channel = 1;
rtp_header->type.Audio.isCNG = false;
}
void Forward(WebRtcRTPHeader* rtp_header) {
++rtp_header->header.sequenceNumber;
rtp_header->header.timestamp += samples_per_packet_;
}
private:
int samples_per_packet_;
uint8_t payload_type_;
};
// This factory method allows access to ACM of a channel, facilitating insertion
// of packets to and pulling audio of ACM.
struct InsertAcm : AudioCodingModuleFactory {
explicit InsertAcm(AudioCodingModule* acm) : acm_(acm) {}
~InsertAcm() {}
virtual AudioCodingModule* Create(int /*id*/) const { return acm_; }
AudioCodingModule* acm_;
};
class VoENetEqStatsTest : public ::testing::Test {
protected:
VoENetEqStatsTest()
: acm1_(new acm1::AudioCodingModuleImpl(1, Clock::GetRealTimeClock())),
acm2_(new acm2::AudioCodingModuleImpl(2)),
voe_(VoiceEngine::Create()),
base_(VoEBase::GetInterface(voe_)),
voe_neteq_stats_(VoENetEqStats::GetInterface(voe_)),
channel_acm1_(-1),
channel_acm2_(-1),
adm_(new FakeAudioDeviceModule),
rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)) {}
~VoENetEqStatsTest() {}
void TearDown() {
voe_neteq_stats_->Release();
base_->DeleteChannel(channel_acm1_);
base_->DeleteChannel(channel_acm2_);
base_->Terminate();
base_->Release();
VoiceEngine::Delete(voe_);
}
void SetUp() {
// Check if all components are valid.
ASSERT_TRUE(voe_ != NULL);
ASSERT_TRUE(base_ != NULL);
ASSERT_TRUE(adm_.get() != NULL);
ASSERT_EQ(0, base_->Init(adm_.get()));
// Set configs.
config_acm1_.Set<AudioCodingModuleFactory>(new InsertAcm(acm1_));
config_acm2_.Set<AudioCodingModuleFactory>(new InsertAcm(acm2_));
// Create channe1s;
channel_acm1_ = base_->CreateChannel(config_acm1_);
ASSERT_NE(-1, channel_acm1_);
channel_acm2_ = base_->CreateChannel(config_acm2_);
ASSERT_NE(-1, channel_acm2_);
CodecInst codec;
AudioCodingModule::Codec("L16", &codec, kSampleRateHz, 1);
codec.pltype = kPayloadType;
// Register L16 codec in ACMs.
ASSERT_EQ(0, acm1_->RegisterReceiveCodec(codec));
ASSERT_EQ(0, acm2_->RegisterReceiveCodec(codec));
rtp_utility_->Populate(&rtp_header_);
}
void InsertPacketAndPullAudio() {
AudioFrame audio_frame;
const uint8_t kPayload[kPayloadSizeBytes] = {0};
ASSERT_EQ(0,
acm1_->IncomingPacket(kPayload, kPayloadSizeBytes, rtp_header_));
ASSERT_EQ(0,
acm2_->IncomingPacket(kPayload, kPayloadSizeBytes, rtp_header_));
ASSERT_EQ(0, acm1_->PlayoutData10Ms(-1, &audio_frame));
ASSERT_EQ(0, acm2_->PlayoutData10Ms(-1, &audio_frame));
rtp_utility_->Forward(&rtp_header_);
}
void JustPullAudio() {
AudioFrame audio_frame;
ASSERT_EQ(0, acm1_->PlayoutData10Ms(-1, &audio_frame));
ASSERT_EQ(0, acm2_->PlayoutData10Ms(-1, &audio_frame));
}
Config config_acm1_;
Config config_acm2_;
// ACMs are inserted into VoE channels, and this class is not the owner of
// them. Therefore, they should not be deleted, not even in destructor.
AudioCodingModule* acm1_;
AudioCodingModule* acm2_;
VoiceEngine* voe_;
VoEBase* base_;
VoENetEqStats* voe_neteq_stats_;
int channel_acm1_;
int channel_acm2_;
scoped_ptr<FakeAudioDeviceModule> adm_;
scoped_ptr<RtpUtility> rtp_utility_;
WebRtcRTPHeader rtp_header_;
};
// Check if the statistics are initialized correctly. Before any call to ACM
// all fields have to be zero.
TEST_F(VoENetEqStatsTest, InitializedToZero) {
AudioDecodingCallStats stats;
ASSERT_EQ(0,
voe_neteq_stats_->GetDecodingCallStatistics(channel_acm1_, &stats));
EXPECT_EQ(0, stats.calls_to_neteq);
EXPECT_EQ(0, stats.calls_to_silence_generator);
EXPECT_EQ(0, stats.decoded_normal);
EXPECT_EQ(0, stats.decoded_cng);
EXPECT_EQ(0, stats.decoded_plc);
EXPECT_EQ(0, stats.decoded_plc_cng);
ASSERT_EQ(0,
voe_neteq_stats_->GetDecodingCallStatistics(channel_acm2_, &stats));
EXPECT_EQ(0, stats.calls_to_neteq);
EXPECT_EQ(0, stats.calls_to_silence_generator);
EXPECT_EQ(0, stats.decoded_normal);
EXPECT_EQ(0, stats.decoded_cng);
EXPECT_EQ(0, stats.decoded_plc);
EXPECT_EQ(0, stats.decoded_plc_cng);
}
// Apply an initial playout delay. Calls to AudioCodingModule::PlayoutData10ms()
// should result in generating silence, check the associated field.
TEST_F(VoENetEqStatsTest, SilenceGeneratorCalled) {
AudioDecodingCallStats stats;
const int kInitialDelay = 100;
acm1_->SetInitialPlayoutDelay(kInitialDelay);
acm2_->SetInitialPlayoutDelay(kInitialDelay);
AudioFrame audio_frame;
int num_calls = 0;
for (int time_ms = 0; time_ms < kInitialDelay;
time_ms += kFrameSizeMs, ++num_calls) {
InsertPacketAndPullAudio();
}
ASSERT_EQ(0,
voe_neteq_stats_->GetDecodingCallStatistics(channel_acm1_, &stats));
EXPECT_EQ(0, stats.calls_to_neteq);
EXPECT_EQ(num_calls, stats.calls_to_silence_generator);
EXPECT_EQ(0, stats.decoded_normal);
EXPECT_EQ(0, stats.decoded_cng);
EXPECT_EQ(0, stats.decoded_plc);
EXPECT_EQ(0, stats.decoded_plc_cng);
ASSERT_EQ(0,
voe_neteq_stats_->GetDecodingCallStatistics(channel_acm2_, &stats));
EXPECT_EQ(0, stats.calls_to_neteq);
EXPECT_EQ(num_calls, stats.calls_to_silence_generator);
EXPECT_EQ(0, stats.decoded_normal);
EXPECT_EQ(0, stats.decoded_cng);
EXPECT_EQ(0, stats.decoded_plc);
EXPECT_EQ(0, stats.decoded_plc_cng);
}
// Insert some packets and pull audio. Check statistics are valid. Then,
// simulate packet loss and check if PLC and PLC-to-CNG statistics are
// correctly updated.
TEST_F(VoENetEqStatsTest, NetEqCalls) {
AudioDecodingCallStats stats;
const int kNumNormalCalls = 10;
AudioFrame audio_frame;
for (int num_calls = 0; num_calls < kNumNormalCalls; ++num_calls) {
InsertPacketAndPullAudio();
}
ASSERT_EQ(0,
voe_neteq_stats_->GetDecodingCallStatistics(channel_acm1_, &stats));
EXPECT_EQ(kNumNormalCalls, stats.calls_to_neteq);
EXPECT_EQ(0, stats.calls_to_silence_generator);
EXPECT_EQ(kNumNormalCalls, stats.decoded_normal);
EXPECT_EQ(0, stats.decoded_cng);
EXPECT_EQ(0, stats.decoded_plc);
EXPECT_EQ(0, stats.decoded_plc_cng);
ASSERT_EQ(0,
voe_neteq_stats_->GetDecodingCallStatistics(channel_acm2_, &stats));
EXPECT_EQ(kNumNormalCalls, stats.calls_to_neteq);
EXPECT_EQ(0, stats.calls_to_silence_generator);
EXPECT_EQ(kNumNormalCalls, stats.decoded_normal);
EXPECT_EQ(0, stats.decoded_cng);
EXPECT_EQ(0, stats.decoded_plc);
EXPECT_EQ(0, stats.decoded_plc_cng);
const int kNumPlc = 3;
const int kNumPlcCng = 5;
// Simulate packet-loss. NetEq first performs PLC then PLC fades to CNG.
for (int n = 0; n < kNumPlc + kNumPlcCng; ++n) {
JustPullAudio();
}
ASSERT_EQ(0,
voe_neteq_stats_->GetDecodingCallStatistics(channel_acm1_, &stats));
EXPECT_EQ(kNumNormalCalls + kNumPlc + kNumPlcCng, stats.calls_to_neteq);
EXPECT_EQ(0, stats.calls_to_silence_generator);
EXPECT_EQ(kNumNormalCalls, stats.decoded_normal);
EXPECT_EQ(0, stats.decoded_cng);
EXPECT_EQ(kNumPlc, stats.decoded_plc);
EXPECT_EQ(kNumPlcCng, stats.decoded_plc_cng);
ASSERT_EQ(0,
voe_neteq_stats_->GetDecodingCallStatistics(channel_acm2_, &stats));
EXPECT_EQ(kNumNormalCalls + kNumPlc + kNumPlcCng, stats.calls_to_neteq);
EXPECT_EQ(0, stats.calls_to_silence_generator);
EXPECT_EQ(kNumNormalCalls, stats.decoded_normal);
EXPECT_EQ(0, stats.decoded_cng);
EXPECT_EQ(kNumPlc, stats.decoded_plc);
EXPECT_EQ(kNumPlcCng, stats.decoded_plc_cng);
}
} // namespace
} // namespace voe
} // namespace webrtc

1
webrtc/voice_engine/voice_engine.gyp
View File

@ -134,6 +134,7 @@
'voe_audio_processing_unittest.cc',
'voe_base_unittest.cc',
'voe_codec_unittest.cc',
'voe_neteq_stats_unittest.cc',
],
'conditions': [
# TODO(henrike): remove build_with_chromium==1 when the bots are