Make VideoSendStreamTest::MaxPacketSize test a whole range of frame sizes, to make sure all corner cases are covered.

BUG=2428
R=pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5229 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/test/configurable_frame_size_encoder.cc

@@ -0,0 +1,93 @@
+/*
+ *  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/test/configurable_frame_size_encoder.h"
+
+#include <string.h>
+
+#include "webrtc/common_video/interface/video_image.h"
+#include "testing/gtest/include/gtest/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+ConfigurableFrameSizeEncoder::ConfigurableFrameSizeEncoder(
+    uint32_t max_frame_size)
+    : callback_(NULL),
+      max_frame_size_(max_frame_size),
+      current_frame_size_(max_frame_size),
+      buffer_(new uint8_t[max_frame_size]) {
+  memset(buffer_.get(), 0, max_frame_size);
+}
+
+ConfigurableFrameSizeEncoder::~ConfigurableFrameSizeEncoder() {}
+
+int32_t ConfigurableFrameSizeEncoder::InitEncode(
+    const VideoCodec* codec_settings,
+    int32_t number_of_cores,
+    uint32_t max_payload_size) {
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t ConfigurableFrameSizeEncoder::Encode(
+    const I420VideoFrame& inputImage,
+    const CodecSpecificInfo* codecSpecificInfo,
+    const std::vector<VideoFrameType>* frame_types) {
+  EncodedImage encodedImage(
+      buffer_.get(), current_frame_size_, max_frame_size_);
+  encodedImage._completeFrame = true;
+  encodedImage._encodedHeight = inputImage.height();
+  encodedImage._encodedWidth = inputImage.width();
+  encodedImage._frameType = kKeyFrame;
+  encodedImage._timeStamp = inputImage.timestamp();
+  encodedImage.capture_time_ms_ = inputImage.render_time_ms();
+  RTPFragmentationHeader* fragmentation = NULL;
+  callback_->Encoded(encodedImage, codecSpecificInfo, fragmentation);
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t ConfigurableFrameSizeEncoder::RegisterEncodeCompleteCallback(
+    EncodedImageCallback* callback) {
+  callback_ = callback;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t ConfigurableFrameSizeEncoder::Release() {
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t ConfigurableFrameSizeEncoder::SetChannelParameters(uint32_t packet_loss,
+                                                           int rtt) {
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t ConfigurableFrameSizeEncoder::SetRates(uint32_t new_bit_rate,
+                                               uint32_t frame_rate) {
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t ConfigurableFrameSizeEncoder::SetPeriodicKeyFrames(bool enable) {
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t ConfigurableFrameSizeEncoder::CodecConfigParameters(uint8_t* buffer,
+                                                            int32_t size) {
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t ConfigurableFrameSizeEncoder::SetFrameSize(uint32_t size) {
+  assert(size <= max_frame_size_);
+  current_frame_size_ = size;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+}  // namespace test
+}  // namespace webrtc

webrtc/test/configurable_frame_size_encoder.h

@@ -0,0 +1,61 @@
+/*
+ *  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_TEST_CONFIGURABLE_FRAME_SIZE_ENCODER_H_
+#define WEBRTC_TEST_CONFIGURABLE_FRAME_SIZE_ENCODER_H_
+
+#include <vector>
+
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+
+namespace webrtc {
+namespace test {
+
+class ConfigurableFrameSizeEncoder : public VideoEncoder {
+ public:
+  explicit ConfigurableFrameSizeEncoder(uint32_t max_frame_size);
+  virtual ~ConfigurableFrameSizeEncoder();
+
+  virtual int32_t InitEncode(const VideoCodec* codec_settings,
+                             int32_t number_of_cores,
+                             uint32_t max_payload_size) OVERRIDE;
+
+  virtual int32_t Encode(const I420VideoFrame& input_image,
+                         const CodecSpecificInfo* codec_specific_info,
+                         const std::vector<VideoFrameType>* frame_types)
+      OVERRIDE;
+
+  virtual int32_t RegisterEncodeCompleteCallback(EncodedImageCallback* callback)
+      OVERRIDE;
+
+  virtual int32_t Release() OVERRIDE;
+
+  virtual int32_t SetChannelParameters(uint32_t packet_loss, int rtt) OVERRIDE;
+
+  virtual int32_t SetRates(uint32_t new_bit_rate, uint32_t frame_rate) OVERRIDE;
+
+  virtual int32_t SetPeriodicKeyFrames(bool enable) OVERRIDE;
+
+  virtual int32_t CodecConfigParameters(uint8_t* buffer, int32_t size) OVERRIDE;
+
+  int32_t SetFrameSize(uint32_t size);
+
+ private:
+  EncodedImageCallback* callback_;
+  const uint32_t max_frame_size_;
+  uint32_t current_frame_size_;
+  scoped_ptr<uint8_t[]> buffer_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // WEBRTC_TEST_CONFIGURABLE_FRAME_SIZE_ENCODER_H_

webrtc/test/webrtc_test_common.gyp

@@ -14,6 +14,8 @@
       'target_name': 'webrtc_test_common',
       'type': 'static_library',
       'sources': [
+        'configurable_frame_size_encoder.cc',
+        'configurable_frame_size_encoder.h',
         'direct_transport.cc',
         'direct_transport.h',
         'fake_audio_device.cc',

webrtc/video/video_send_stream_tests.cc

@@ -24,6 +24,7 @@
 #include "webrtc/system_wrappers/interface/thread_wrapper.h"
 #include "webrtc/test/direct_transport.h"
 #include "webrtc/test/fake_encoder.h"
+#include "webrtc/test/configurable_frame_size_encoder.h"
 #include "webrtc/test/frame_generator_capturer.h"
 #include "webrtc/test/null_transport.h"
 #include "webrtc/test/rtp_rtcp_observer.h"
@@ -528,11 +529,23 @@ TEST_F(VideoSendStreamTest, RetransmitsNackOverRtxWithPacing) {
   TestNackRetransmission(kSendRtxSsrc, kSendRtxPayloadType, true);
 }
 
-TEST_F(VideoSendStreamTest, MaxPacketSize) {
-  class PacketSizeObserver : public test::RtpRtcpObserver {
+TEST_F(VideoSendStreamTest, FragmentsAccordingToMaxPacketSize) {
+  // Observer that verifies that the expected number of packets and bytes
+  // arrive for each frame size, then increases frame size by one until
+  // stop_size has been reached.
+  class FrameFragmentationObserver : public test::RtpRtcpObserver {
    public:
-    PacketSizeObserver(size_t max_length) : RtpRtcpObserver(30 * 1000),
-      max_length_(max_length), accumulated_size_(0) {}
+    FrameFragmentationObserver(size_t max_length,
+                       uint32_t start_size,
+                       uint32_t stop_size,
+                       test::ConfigurableFrameSizeEncoder* encoder)
+        : RtpRtcpObserver(30 * 1000),
+          max_length_(max_length),
+          accumulated_size_(0),
+          accumulated_payload_(0),
+          stop_size_(stop_size),
+          current_size_(start_size),
+          encoder_(encoder) {}
 
     virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
       RTPHeader header;
@@ -542,15 +555,25 @@ TEST_F(VideoSendStreamTest, MaxPacketSize) {
       EXPECT_LE(length, max_length_);
 
       accumulated_size_ += length;
+      // Payload size = packet size - minus RTP header, padding and one byte
+      // generic header.
+      accumulated_payload_ +=
+          length - (header.headerLength + header.paddingLength + 1);
 
-      // Marker bit set indicates last fragment of a packet
+      // Marker bit set indicates last packet of a frame.
       if (header.markerBit) {
-        if (accumulated_size_ + length > max_length_) {
-          // The packet was fragmented, total size was larger than max size,
-          // but size of individual fragments were within size limit => pass!
+        EXPECT_GE(accumulated_size_, current_size_);
+        EXPECT_EQ(accumulated_payload_, current_size_);
+
+        if (current_size_ == stop_size_) {
+          EXPECT_GE(current_size_, max_length_);
           observation_complete_->Set();
+        } else {
+          // Last packet of frame; reset and increase frame size.
+          accumulated_size_ = 0;
+          accumulated_payload_ = 0;
+          encoder_->SetFrameSize(++current_size_);
         }
-        accumulated_size_ = 0; // Last fragment, reset packet size
       }
 
       return SEND_PACKET;
@@ -559,15 +582,28 @@ TEST_F(VideoSendStreamTest, MaxPacketSize) {
    private:
     size_t max_length_;
     size_t accumulated_size_;
+    size_t accumulated_payload_;
+
+    uint32_t stop_size_;
+    uint32_t current_size_;
+    test::ConfigurableFrameSizeEncoder* encoder_;
   };
 
+  // Use a fake encoder to output a frame of every size in the range [90, 290],
+  // for each size making sure that the exact number of payload bytes received
+  // is correct and that packets are fragmented to respect max packet size.
   static const uint32_t kMaxPacketSize = 128;
+  static const uint32_t start = 90;
+  static const uint32_t stop = 290;
 
-  PacketSizeObserver observer(kMaxPacketSize);
+  test::ConfigurableFrameSizeEncoder encoder(stop);
+  encoder.SetFrameSize(start);
+  FrameFragmentationObserver observer(kMaxPacketSize, start, stop, &encoder);
   Call::Config call_config(observer.SendTransport());
   scoped_ptr<Call> call(Call::Create(call_config));
 
   VideoSendStream::Config send_config = GetSendTestConfig(call.get(), 1);
+  send_config.encoder = &encoder;
   send_config.rtp.max_packet_size = kMaxPacketSize;
 
   RunSendTest(call.get(), send_config, &observer);