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| author | Android Chromium Automerger <chromium-automerger@android> | 2014-07-09 15:00:07 +0000 |
|---|---|---|
| committer | Android Chromium Automerger <chromium-automerger@android> | 2014-07-09 15:00:07 +0000 |
| commit | 10b9861436e47694b29da87b7890287041a95a74 (patch) | |
| tree | 64d09be0638cccd20e1ca345ff522372b708ab67 | |
| parent | 3ded5808c2dec22ec97b3b0c855f1bcb75c1db09 (diff) | |
| parent | 138adbb0bcdab60afda25a8727e5a071abc4ae36 (diff) | |
| download | webrtc-10b9861436e47694b29da87b7890287041a95a74.tar.gz | |
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 138adbb0bcdab60afda25a8727e5a071abc4ae36
This commit was generated by merge_from_chromium.py.
Change-Id: Iffa5413ebfb78de36b84b4e85d94adc093f912df
147 files changed, 3713 insertions, 1866 deletions
diff --git a/base/base.gyp b/base/base.gyp index f05c3251..f9ba4041 100644 --- a/base/base.gyp +++ b/base/base.gyp @@ -711,7 +711,7 @@ 'scoped_autorelease_pool.mm', ], }], - ['OS=="ios" or os_posix==0', { + ['OS=="ios"', { 'sources!': [ 'openssl.h', 'openssladapter.cc', diff --git a/base/timing.cc b/base/timing.cc index aa1fc429..0c5ed5e1 100644 --- a/base/timing.cc +++ b/base/timing.cc @@ -46,6 +46,7 @@ Timing::~Timing() { #endif } +// static double Timing::WallTimeNow() { #if defined(WEBRTC_POSIX) struct timeval time; diff --git a/base/timing.h b/base/timing.h index 58b17a9f..1dee6076 100644 --- a/base/timing.h +++ b/base/timing.h @@ -24,7 +24,8 @@ class Timing { // WallTimeNow() returns the current wall-clock time in seconds, // within 10 milliseconds resolution. - virtual double WallTimeNow(); + // WallTimeNow is static and does not require a timer_handle_ on Windows. + static double WallTimeNow(); // TimerNow() is like WallTimeNow(), but is monotonically // increasing. It returns seconds in resolution of 10 microseconds diff --git a/build/tsan_suppressions.cc b/build/tsan_suppressions.cc index 4002d3c2..02e13af2 100644 --- a/build/tsan_suppressions.cc +++ b/build/tsan_suppressions.cc @@ -23,9 +23,11 @@ char kTSanDefaultSuppressions[] = // WebRTC specific suppressions. -// False positive in system wrappers. -// https://code.google.com/p/webrtc/issues/detail?id=300 -"race:webrtc/system_wrappers/source/thread_posix.cc\n" +// Usage of trace callback and trace level is racy in libjingle_media_unittests. +// https://code.google.com/p/webrtc/issues/detail?id=3372 +"race:webrtc::TraceImpl::WriteToFile\n" +"race:webrtc::VideoEngine::SetTraceFilter\n" +"race:webrtc::VoiceEngine::SetTraceFilter\n" // Audio processing // https://code.google.com/p/webrtc/issues/detail?id=2521 for details. @@ -65,7 +67,6 @@ char kTSanDefaultSuppressions[] = "deadlock:webrtc::ProcessThreadImpl::RegisterModule\n" "deadlock:webrtc::RTCPReceiver::SetSsrcs\n" "deadlock:webrtc::RTPSenderAudio::RegisterAudioPayload\n" -"deadlock:webrtc/system_wrappers/source/logging_unittest.cc\n" "deadlock:webrtc::test::UdpSocketManagerPosixImpl::RemoveSocket\n" "deadlock:webrtc::vcm::VideoReceiver::RegisterPacketRequestCallback\n" "deadlock:webrtc::VideoSendStreamTest_SuspendBelowMinBitrate_Test::TestBody\n" diff --git a/build/webrtc.gni b/build/webrtc.gni index 409ae1d2..fcbf04f7 100644 --- a/build/webrtc.gni +++ b/build/webrtc.gni @@ -25,6 +25,9 @@ declare_args() { # when building voice engine exclusively. enable_video = true + # Disable this to not build libvpx and instead use an externally provided lib. + build_libvpx = true + # Selects fixed-point code where possible. prefer_fixed_point = false @@ -32,10 +35,16 @@ declare_args() { # which can be easily parsed for offline processing. enable_data_logging = false - build_libjpeg = true # Enables the use of protocol buffers for debug recordings. enable_protobuf = true + # Disable these to not build components which can be externally provided. + build_json = true + build_libjpeg = true + build_libyuv = true + build_libvpx = true + build_ssl = true + # Disable by default. have_dbus_glib = false @@ -86,8 +86,6 @@ class Call { static Call* Create(const Call::Config& config, const webrtc::Config& webrtc_config); - virtual VideoSendStream::Config GetDefaultSendConfig() = 0; - virtual VideoSendStream* CreateVideoSendStream( const VideoSendStream::Config& config, const std::vector<VideoStream>& video_streams, @@ -95,8 +93,6 @@ class Call { virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0; - virtual VideoReceiveStream::Config GetDefaultReceiveConfig() = 0; - virtual VideoReceiveStream* CreateVideoReceiveStream( const VideoReceiveStream::Config& config) = 0; virtual void DestroyVideoReceiveStream( diff --git a/common_audio/signal_processing/include/signal_processing_library.h b/common_audio/signal_processing/include/signal_processing_library.h index 3a5d51cc..a658645d 100644 --- a/common_audio/signal_processing/include/signal_processing_library.h +++ b/common_audio/signal_processing/include/signal_processing_library.h @@ -111,8 +111,6 @@ // Shifting with negative numbers allowed // Positive means left shift -#define WEBRTC_SPL_SHIFT_W16(x, c) \ - (((c) >= 0) ? ((x) << (c)) : ((x) >> (-(c)))) #define WEBRTC_SPL_SHIFT_W32(x, c) \ (((c) >= 0) ? ((x) << (c)) : ((x) >> (-(c)))) diff --git a/common_audio/signal_processing/signal_processing_unittest.cc b/common_audio/signal_processing/signal_processing_unittest.cc index 81ca3694..a68840e8 100644 --- a/common_audio/signal_processing/signal_processing_unittest.cc +++ b/common_audio/signal_processing/signal_processing_unittest.cc @@ -79,7 +79,6 @@ TEST_F(SplTest, MacroTest) { // Shifting with negative numbers allowed int shift_amount = 1; // Workaround compiler warning using variable here. // Positive means left shift - EXPECT_EQ(32766, WEBRTC_SPL_SHIFT_W16(a, shift_amount)); EXPECT_EQ(32766, WEBRTC_SPL_SHIFT_W32(a, shift_amount)); // Shifting with negative numbers not allowed diff --git a/common_types.h b/common_types.h index 6892a83f..193e2818 100644 --- a/common_types.h +++ b/common_types.h @@ -587,10 +587,24 @@ struct VideoCodecVP8 { } }; +// H264 specific. +struct VideoCodecH264 +{ + VideoCodecProfile profile; + bool frameDroppingOn; + int keyFrameInterval; + // These are NULL/0 if not externally negotiated. + const uint8_t* spsData; + size_t spsLen; + const uint8_t* ppsData; + size_t ppsLen; +}; + // Video codec types enum VideoCodecType { kVideoCodecVP8, + kVideoCodecH264, kVideoCodecI420, kVideoCodecRED, kVideoCodecULPFEC, @@ -601,6 +615,7 @@ enum VideoCodecType union VideoCodecUnion { VideoCodecVP8 VP8; + VideoCodecH264 H264; }; diff --git a/common_video/BUILD.gn b/common_video/BUILD.gn index 2e877e6b..69915e34 100644 --- a/common_video/BUILD.gn +++ b/common_video/BUILD.gn @@ -8,6 +8,38 @@ import("../build/webrtc.gni") +config("common_video_config") { + include_dirs = [ + "interface", + "libyuv/include", + ] +} + source_set("common_video") { - # TODO(pbos): Implement. + sources = [ + "i420_video_frame.cc", + "interface/i420_video_frame.h", + "interface/native_handle.h", + "interface/texture_video_frame.h", + "libyuv/include/scaler.h", + "libyuv/include/webrtc_libyuv.h", + "libyuv/scaler.cc", + "libyuv/webrtc_libyuv.cc", + "plane.cc", + "plane.h", + "texture_video_frame.cc" + ] + + include_dirs = [ "../modules/interface" ] + + direct_dependent_configs = [ ":common_video_config" ] + + deps = [ "../system_wrappers" ] + + if (build_libyuv) { + deps += [ "//third_party/libyuv" ] + } else { + # Need to add a directory normally exported by libyuv. + include_dirs += [ "//third_party/libyuv/include" ] + } } diff --git a/engine_configurations.h b/engine_configurations.h index be858b8e..e9f23097 100644 --- a/engine_configurations.h +++ b/engine_configurations.h @@ -51,6 +51,7 @@ #define VIDEOCODEC_I420 #define VIDEOCODEC_VP8 +#define VIDEOCODEC_H264 // ============================================================================ // VoiceEngine diff --git a/examples/android/media_demo/jni/video_engine_jni.cc b/examples/android/media_demo/jni/video_engine_jni.cc index c6589ece..c7af1c54 100644 --- a/examples/android/media_demo/jni/video_engine_jni.cc +++ b/examples/android/media_demo/jni/video_engine_jni.cc @@ -710,3 +710,9 @@ JOWW(void, VideoCodecInst_setMaxFrameRate)(JNIEnv* jni, jobject j_codec, JOWW(void, CameraDesc_dispose)(JNIEnv* jni, jobject j_camera) { delete GetCameraDesc(jni, j_camera); } + +JOWW(jint, VideoEngine_setLocalSSRC)(JNIEnv* jni, jobject j_vie, jint channel, + jint ssrc) { + VideoEngineData* vie_data = GetVideoEngineData(jni, j_vie); + return vie_data->rtp->SetLocalSSRC(channel, ssrc); +} diff --git a/examples/android/media_demo/src/org/webrtc/webrtcdemo/MediaEngine.java b/examples/android/media_demo/src/org/webrtc/webrtcdemo/MediaEngine.java index f6284315..f55bcccf 100644 --- a/examples/android/media_demo/src/org/webrtc/webrtcdemo/MediaEngine.java +++ b/examples/android/media_demo/src/org/webrtc/webrtcdemo/MediaEngine.java @@ -259,7 +259,7 @@ public class MediaEngine implements VideoDecodeEncodeObserver { public void setTrace(boolean enable) { if (enable) { vie.setTraceFile("/sdcard/trace.txt", false); - vie.setTraceFilter(VideoEngine.TraceLevel.TRACE_ERROR); + vie.setTraceFilter(VideoEngine.TraceLevel.TRACE_ALL); return; } vie.setTraceFilter(VideoEngine.TraceLevel.TRACE_NONE); @@ -561,6 +561,14 @@ public class MediaEngine implements VideoDecodeEncodeObserver { check(vie.setSendDestination(videoChannel, videoTxPort, remoteIp) == 0, "Failed setSendDestination"); } + + // Setting localSSRC manually (arbitrary value) for loopback test, + // As otherwise we will get a clash and a new SSRC will be set, + // Which will reset the receiver and other minor issues. + if (remoteIp.equals("127.0.0.1")) { + check(vie.setLocalSSRC(videoChannel, 0x01234567) == 0, + "Failed setLocalSSRC"); + } } public int videoTxPort() { diff --git a/examples/android/media_demo/src/org/webrtc/webrtcdemo/VideoEngine.java b/examples/android/media_demo/src/org/webrtc/webrtcdemo/VideoEngine.java index 7cfa5856..677ff42f 100644 --- a/examples/android/media_demo/src/org/webrtc/webrtcdemo/VideoEngine.java +++ b/examples/android/media_demo/src/org/webrtc/webrtcdemo/VideoEngine.java @@ -116,4 +116,5 @@ public class VideoEngine { return stopRtpDump(channel, direction.ordinal()); } private native int stopRtpDump(int channel, int direction); + public native int setLocalSSRC(int channel, int ssrc); } diff --git a/experiments.h b/experiments.h index b03d248c..3b019b48 100644 --- a/experiments.h +++ b/experiments.h @@ -14,16 +14,6 @@ #include "webrtc/typedefs.h" namespace webrtc { -struct PaddingStrategy { - PaddingStrategy() - : redundant_payloads(false) {} - explicit PaddingStrategy(bool redundant_payloads) - : redundant_payloads(redundant_payloads) {} - virtual ~PaddingStrategy() {} - - const bool redundant_payloads; -}; - struct RemoteBitrateEstimatorMinRate { RemoteBitrateEstimatorMinRate() : min_rate(30000) {} RemoteBitrateEstimatorMinRate(uint32_t min_rate) : min_rate(min_rate) {} diff --git a/modules/audio_device/dummy/file_audio_device_factory.cc b/modules/audio_device/dummy/file_audio_device_factory.cc index db35bf11..5e252305 100644 --- a/modules/audio_device/dummy/file_audio_device_factory.cc +++ b/modules/audio_device/dummy/file_audio_device_factory.cc @@ -32,12 +32,19 @@ FileAudioDevice* FileAudioDeviceFactory::CreateFileAudioDevice( void FileAudioDeviceFactory::SetFilenamesToUse( const char* inputAudioFilename, const char* outputAudioFilename) { +#ifdef WEBRTC_DUMMY_FILE_DEVICES assert(strlen(inputAudioFilename) < MAX_FILENAME_LEN && strlen(outputAudioFilename) < MAX_FILENAME_LEN); // Copy the strings since we don't know the lifetime of the input pointers. strncpy(_inputAudioFilename, inputAudioFilename, MAX_FILENAME_LEN); strncpy(_outputAudioFilename, outputAudioFilename, MAX_FILENAME_LEN); +#else + // Sanity: must be compiled with the right define to run this. + printf("Trying to use dummy file devices, but is not compiled " + "with WEBRTC_DUMMY_FILE_DEVICES. Bailing out.\n"); + exit(1); +#endif } } // namespace webrtc diff --git a/modules/audio_processing/aec/aec_rdft_neon.c b/modules/audio_processing/aec/aec_rdft_neon.c index 80892d26..a9c79b7b 100644 --- a/modules/audio_processing/aec/aec_rdft_neon.c +++ b/modules/audio_processing/aec/aec_rdft_neon.c @@ -178,8 +178,96 @@ static void cftmdl_128_neon(float* a) { } } +__inline static float32x4_t reverse_order_f32x4(float32x4_t in) { + // A B C D -> C D A B + const float32x4_t rev = vcombine_f32(vget_high_f32(in), vget_low_f32(in)); + // C D A B -> D C B A + return vrev64q_f32(rev); +} + +static void rftfsub_128_neon(float* a) { + const float* c = rdft_w + 32; + int j1, j2, k1, k2; + float wkr, wki, xr, xi, yr, yi; + const float32x4_t mm_half = vdupq_n_f32(0.5f); + + // Vectorized code (four at once). + // Note: commented number are indexes for the first iteration of the loop. + for (j1 = 1, j2 = 2; j2 + 7 < 64; j1 += 4, j2 += 8) { + // Load 'wk'. + const float32x4_t c_j1 = vld1q_f32(&c[j1]); // 1, 2, 3, 4, + const float32x4_t c_k1 = vld1q_f32(&c[29 - j1]); // 28, 29, 30, 31, + const float32x4_t wkrt = vsubq_f32(mm_half, c_k1); // 28, 29, 30, 31, + const float32x4_t wkr_ = reverse_order_f32x4(wkrt); + const float32x4_t wki_ = c_j1; // 1, 2, 3, 4, + // Load and shuffle 'a'. + // 2, 4, 6, 8, 3, 5, 7, 9 + float32x4x2_t a_j2_p = vld2q_f32(&a[0 + j2]); + // 120, 122, 124, 126, 121, 123, 125, 127, + const float32x4x2_t k2_0_4 = vld2q_f32(&a[122 - j2]); + // 126, 124, 122, 120 + const float32x4_t a_k2_p0 = reverse_order_f32x4(k2_0_4.val[0]); + // 127, 125, 123, 121 + const float32x4_t a_k2_p1 = reverse_order_f32x4(k2_0_4.val[1]); + // Calculate 'x'. + const float32x4_t xr_ = vsubq_f32(a_j2_p.val[0], a_k2_p0); + // 2-126, 4-124, 6-122, 8-120, + const float32x4_t xi_ = vaddq_f32(a_j2_p.val[1], a_k2_p1); + // 3-127, 5-125, 7-123, 9-121, + // Calculate product into 'y'. + // yr = wkr * xr - wki * xi; + // yi = wkr * xi + wki * xr; + const float32x4_t a_ = vmulq_f32(wkr_, xr_); + const float32x4_t b_ = vmulq_f32(wki_, xi_); + const float32x4_t c_ = vmulq_f32(wkr_, xi_); + const float32x4_t d_ = vmulq_f32(wki_, xr_); + const float32x4_t yr_ = vsubq_f32(a_, b_); // 2-126, 4-124, 6-122, 8-120, + const float32x4_t yi_ = vaddq_f32(c_, d_); // 3-127, 5-125, 7-123, 9-121, + // Update 'a'. + // a[j2 + 0] -= yr; + // a[j2 + 1] -= yi; + // a[k2 + 0] += yr; + // a[k2 + 1] -= yi; + // 126, 124, 122, 120, + const float32x4_t a_k2_p0n = vaddq_f32(a_k2_p0, yr_); + // 127, 125, 123, 121, + const float32x4_t a_k2_p1n = vsubq_f32(a_k2_p1, yi_); + // Shuffle in right order and store. + const float32x4_t a_k2_p0nr = vrev64q_f32(a_k2_p0n); + const float32x4_t a_k2_p1nr = vrev64q_f32(a_k2_p1n); + // 124, 125, 126, 127, 120, 121, 122, 123 + const float32x4x2_t a_k2_n = vzipq_f32(a_k2_p0nr, a_k2_p1nr); + // 2, 4, 6, 8, + a_j2_p.val[0] = vsubq_f32(a_j2_p.val[0], yr_); + // 3, 5, 7, 9, + a_j2_p.val[1] = vsubq_f32(a_j2_p.val[1], yi_); + // 2, 3, 4, 5, 6, 7, 8, 9, + vst2q_f32(&a[0 + j2], a_j2_p); + + vst1q_f32(&a[122 - j2], a_k2_n.val[1]); + vst1q_f32(&a[126 - j2], a_k2_n.val[0]); + } + + // Scalar code for the remaining items. + for (; j2 < 64; j1 += 1, j2 += 2) { + k2 = 128 - j2; + k1 = 32 - j1; + wkr = 0.5f - c[k1]; + wki = c[j1]; + xr = a[j2 + 0] - a[k2 + 0]; + xi = a[j2 + 1] + a[k2 + 1]; + yr = wkr * xr - wki * xi; + yi = wkr * xi + wki * xr; + a[j2 + 0] -= yr; + a[j2 + 1] -= yi; + a[k2 + 0] += yr; + a[k2 + 1] -= yi; + } +} + void aec_rdft_init_neon(void) { cft1st_128 = cft1st_128_neon; cftmdl_128 = cftmdl_128_neon; + rftfsub_128 = rftfsub_128_neon; } diff --git a/modules/audio_processing/aec/aec_resampler.c b/modules/audio_processing/aec/aec_resampler.c index 5382665e..469b8111 100644 --- a/modules/audio_processing/aec/aec_resampler.c +++ b/modules/audio_processing/aec/aec_resampler.c @@ -26,7 +26,7 @@ enum { }; typedef struct { - short buffer[kResamplerBufferSize]; + float buffer[kResamplerBufferSize]; float position; int deviceSampleRateHz; @@ -71,15 +71,15 @@ int WebRtcAec_FreeResampler(void* resampInst) { } void WebRtcAec_ResampleLinear(void* resampInst, - const short* inspeech, + const float* inspeech, int size, float skew, - short* outspeech, + float* outspeech, int* size_out) { resampler_t* obj = (resampler_t*)resampInst; - short* y; - float be, tnew, interp; + float* y; + float be, tnew; int tn, mm; assert(!(size < 0 || size > 2 * FRAME_LEN)); @@ -91,7 +91,7 @@ void WebRtcAec_ResampleLinear(void* resampInst, // Add new frame data in lookahead memcpy(&obj->buffer[FRAME_LEN + kResamplingDelay], inspeech, - size * sizeof(short)); + size * sizeof(inspeech[0])); // Sample rate ratio be = 1 + skew; @@ -106,15 +106,7 @@ void WebRtcAec_ResampleLinear(void* resampInst, while (tn < size) { // Interpolation - interp = y[tn] + (tnew - tn) * (y[tn + 1] - y[tn]); - - if (interp > 32767) { - interp = 32767; - } else if (interp < -32768) { - interp = -32768; - } - - outspeech[mm] = (short)interp; + outspeech[mm] = y[tn] + (tnew - tn) * (y[tn + 1] - y[tn]); mm++; tnew = be * mm + obj->position; @@ -127,7 +119,7 @@ void WebRtcAec_ResampleLinear(void* resampInst, // Shift buffer memmove(obj->buffer, &obj->buffer[size], - (kResamplerBufferSize - size) * sizeof(short)); + (kResamplerBufferSize - size) * sizeof(obj->buffer[0])); } int WebRtcAec_GetSkew(void* resampInst, int rawSkew, float* skewEst) { diff --git a/modules/audio_processing/aec/aec_resampler.h b/modules/audio_processing/aec/aec_resampler.h index e42c056f..73e28212 100644 --- a/modules/audio_processing/aec/aec_resampler.h +++ b/modules/audio_processing/aec/aec_resampler.h @@ -30,10 +30,10 @@ int WebRtcAec_GetSkew(void* resampInst, int rawSkew, float* skewEst); // Resamples input using linear interpolation. void WebRtcAec_ResampleLinear(void* resampInst, - const short* inspeech, + const float* inspeech, int size, float skew, - short* outspeech, + float* outspeech, int* size_out); #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_ diff --git a/modules/audio_processing/aec/echo_cancellation.c b/modules/audio_processing/aec/echo_cancellation.c index ba3b9243..b58edcbb 100644 --- a/modules/audio_processing/aec/echo_cancellation.c +++ b/modules/audio_processing/aec/echo_cancellation.c @@ -294,17 +294,12 @@ int32_t WebRtcAec_Init(void* aecInst, int32_t sampFreq, int32_t scSampFreq) { // only buffer L band for farend int32_t WebRtcAec_BufferFarend(void* aecInst, - const int16_t* farend, + const float* farend, int16_t nrOfSamples) { aecpc_t* aecpc = aecInst; - int32_t retVal = 0; int newNrOfSamples = (int)nrOfSamples; - short newFarend[MAX_RESAMP_LEN]; - const int16_t* farend_ptr = farend; - float tmp_farend[MAX_RESAMP_LEN]; - const float* farend_float = tmp_farend; - float skew; - int i = 0; + float new_farend[MAX_RESAMP_LEN]; + const float* farend_ptr = farend; if (farend == NULL) { aecpc->lastError = AEC_NULL_POINTER_ERROR; @@ -322,17 +317,15 @@ int32_t WebRtcAec_BufferFarend(void* aecInst, return -1; } - skew = aecpc->skew; - if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) { // Resample and get a new number of samples WebRtcAec_ResampleLinear(aecpc->resampler, farend, nrOfSamples, - skew, - newFarend, + aecpc->skew, + new_farend, &newNrOfSamples); - farend_ptr = (const int16_t*)newFarend; + farend_ptr = new_farend; } aecpc->farend_started = 1; @@ -343,32 +336,31 @@ int32_t WebRtcAec_BufferFarend(void* aecInst, WebRtc_WriteBuffer( aecpc->far_pre_buf_s16, farend_ptr, (size_t)newNrOfSamples); #endif - // Cast to float and write the time-domain data to |far_pre_buf|. - for (i = 0; i < newNrOfSamples; i++) { - tmp_farend[i] = (float)farend_ptr[i]; - } - WebRtc_WriteBuffer(aecpc->far_pre_buf, farend_float, (size_t)newNrOfSamples); + // Write the time-domain data to |far_pre_buf|. + WebRtc_WriteBuffer(aecpc->far_pre_buf, farend_ptr, (size_t)newNrOfSamples); // Transform to frequency domain if we have enough data. while (WebRtc_available_read(aecpc->far_pre_buf) >= PART_LEN2) { // We have enough data to pass to the FFT, hence read PART_LEN2 samples. - WebRtc_ReadBuffer( - aecpc->far_pre_buf, (void**)&farend_float, tmp_farend, PART_LEN2); - - WebRtcAec_BufferFarendPartition(aecpc->aec, farend_float); + { + float* ptmp; + float tmp[PART_LEN2]; + WebRtc_ReadBuffer(aecpc->far_pre_buf, (void**)&ptmp, tmp, PART_LEN2); + WebRtcAec_BufferFarendPartition(aecpc->aec, ptmp); + } // Rewind |far_pre_buf| PART_LEN samples for overlap before continuing. WebRtc_MoveReadPtr(aecpc->far_pre_buf, -PART_LEN); #ifdef WEBRTC_AEC_DEBUG_DUMP WebRtc_ReadBuffer( - aecpc->far_pre_buf_s16, (void**)&farend_ptr, newFarend, PART_LEN2); + aecpc->far_pre_buf_s16, (void**)&farend_ptr, new_farend, PART_LEN2); WebRtc_WriteBuffer( WebRtcAec_far_time_buf(aecpc->aec), &farend_ptr[PART_LEN], 1); WebRtc_MoveReadPtr(aecpc->far_pre_buf_s16, -PART_LEN); #endif } - return retVal; + return 0; } int32_t WebRtcAec_Process(void* aecInst, diff --git a/modules/audio_processing/aec/include/echo_cancellation.h b/modules/audio_processing/aec/include/echo_cancellation.h index dc64a345..0cf6a5a4 100644 --- a/modules/audio_processing/aec/include/echo_cancellation.h +++ b/modules/audio_processing/aec/include/echo_cancellation.h @@ -114,7 +114,7 @@ int32_t WebRtcAec_Init(void* aecInst, int32_t sampFreq, int32_t scSampFreq); * Inputs Description * ------------------------------------------------------------------- * void* aecInst Pointer to the AEC instance - * int16_t* farend In buffer containing one frame of + * const float* farend In buffer containing one frame of * farend signal for L band * int16_t nrOfSamples Number of samples in farend buffer * @@ -124,7 +124,7 @@ int32_t WebRtcAec_Init(void* aecInst, int32_t sampFreq, int32_t scSampFreq); * -1: error */ int32_t WebRtcAec_BufferFarend(void* aecInst, - const int16_t* farend, + const float* farend, int16_t nrOfSamples); /* diff --git a/modules/audio_processing/aec/system_delay_unittest.cc b/modules/audio_processing/aec/system_delay_unittest.cc index 5fbc5600..f81ce476 100644 --- a/modules/audio_processing/aec/system_delay_unittest.cc +++ b/modules/audio_processing/aec/system_delay_unittest.cc @@ -47,7 +47,7 @@ class SystemDelayTest : public ::testing::Test { int samples_per_frame_; // Dummy input/output speech data. static const int kSamplesPerChunk = 160; - int16_t far_[kSamplesPerChunk]; + float far_[kSamplesPerChunk]; float near_[kSamplesPerChunk]; float out_[kSamplesPerChunk]; }; @@ -55,9 +55,10 @@ class SystemDelayTest : public ::testing::Test { SystemDelayTest::SystemDelayTest() : handle_(NULL), self_(NULL), samples_per_frame_(0) { // Dummy input data are set with more or less arbitrary non-zero values. - memset(far_, 1, sizeof(far_)); - for (int i = 0; i < kSamplesPerChunk; i++) + for (int i = 0; i < kSamplesPerChunk; i++) { + far_[i] = 257.0; near_[i] = 514.0; + } memset(out_, 0, sizeof(out_)); } diff --git a/modules/audio_processing/aecm/aecm_core_c.c b/modules/audio_processing/aecm/aecm_core_c.c index f8491e97..00323ccd 100644 --- a/modules/audio_processing/aecm/aecm_core_c.c +++ b/modules/audio_processing/aecm/aecm_core_c.c @@ -328,7 +328,7 @@ int WebRtcAecm_ProcessBlock(AecmCore_t * aecm, int16_t zeros32, zeros16; int16_t zerosDBufNoisy, zerosDBufClean, zerosXBuf; int far_q; - int16_t resolutionDiff, qDomainDiff; + int16_t resolutionDiff, qDomainDiff, dfa_clean_q_domain_diff; const int kMinPrefBand = 4; const int kMaxPrefBand = 24; @@ -499,29 +499,28 @@ int WebRtcAecm_ProcessBlock(AecmCore_t * aecm, } zeros16 = WebRtcSpl_NormW16(aecm->nearFilt[i]); - if ((zeros16 < (aecm->dfaCleanQDomain - aecm->dfaCleanQDomainOld)) - & (aecm->nearFilt[i])) - { - tmp16no1 = WEBRTC_SPL_SHIFT_W16(aecm->nearFilt[i], zeros16); - qDomainDiff = zeros16 - aecm->dfaCleanQDomain + aecm->dfaCleanQDomainOld; - } else - { - tmp16no1 = WEBRTC_SPL_SHIFT_W16(aecm->nearFilt[i], - aecm->dfaCleanQDomain - - aecm->dfaCleanQDomainOld); + assert(zeros16 >= 0); // |zeros16| is a norm, hence non-negative. + dfa_clean_q_domain_diff = aecm->dfaCleanQDomain - aecm->dfaCleanQDomainOld; + if (zeros16 < dfa_clean_q_domain_diff && aecm->nearFilt[i]) { + tmp16no1 = aecm->nearFilt[i] << zeros16; + qDomainDiff = zeros16 - dfa_clean_q_domain_diff; + tmp16no2 = ptrDfaClean[i] >> -qDomainDiff; + } else { + tmp16no1 = dfa_clean_q_domain_diff < 0 + ? aecm->nearFilt[i] >> -dfa_clean_q_domain_diff + : aecm->nearFilt[i] << dfa_clean_q_domain_diff; qDomainDiff = 0; + tmp16no2 = ptrDfaClean[i]; } - tmp16no2 = WEBRTC_SPL_SHIFT_W16(ptrDfaClean[i], qDomainDiff); tmp32no1 = (int32_t)(tmp16no2 - tmp16no1); tmp16no2 = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32no1, 4); tmp16no2 += tmp16no1; zeros16 = WebRtcSpl_NormW16(tmp16no2); - if ((tmp16no2) & (-qDomainDiff > zeros16)) - { + if ((tmp16no2) & (-qDomainDiff > zeros16)) { aecm->nearFilt[i] = WEBRTC_SPL_WORD16_MAX; - } else - { - aecm->nearFilt[i] = WEBRTC_SPL_SHIFT_W16(tmp16no2, -qDomainDiff); + } else { + aecm->nearFilt[i] = qDomainDiff < 0 ? tmp16no2 << -qDomainDiff + : tmp16no2 >> qDomainDiff; } // Wiener filter coefficients, resulting hnl in Q14 diff --git a/modules/audio_processing/aecm/aecm_core_mips.c b/modules/audio_processing/aecm/aecm_core_mips.c index 6a231b38..31f232b8 100644 --- a/modules/audio_processing/aecm/aecm_core_mips.c +++ b/modules/audio_processing/aecm/aecm_core_mips.c @@ -836,7 +836,7 @@ int WebRtcAecm_ProcessBlock(AecmCore_t* aecm, int16_t zeros32, zeros16; int16_t zerosDBufNoisy, zerosDBufClean, zerosXBuf; int far_q; - int16_t resolutionDiff, qDomainDiff; + int16_t resolutionDiff, qDomainDiff, dfa_clean_q_domain_diff; const int kMinPrefBand = 4; const int kMaxPrefBand = 24; @@ -1002,15 +1002,16 @@ int WebRtcAecm_ProcessBlock(AecmCore_t* aecm, } zeros16 = WebRtcSpl_NormW16(aecm->nearFilt[i]); - if ((zeros16 < (aecm->dfaCleanQDomain - aecm->dfaCleanQDomainOld)) - & (aecm->nearFilt[i])) { - tmp16no1 = WEBRTC_SPL_SHIFT_W16(aecm->nearFilt[i], zeros16); - qDomainDiff = zeros16 - aecm->dfaCleanQDomain + aecm->dfaCleanQDomainOld; - tmp16no2 = WEBRTC_SPL_SHIFT_W16(ptrDfaClean[i], qDomainDiff); + assert(zeros16 >= 0); // |zeros16| is a norm, hence non-negative. + dfa_clean_q_domain_diff = aecm->dfaCleanQDomain - aecm->dfaCleanQDomainOld; + if (zeros16 < dfa_clean_q_domain_diff && aecm->nearFilt[i]) { + tmp16no1 = aecm->nearFilt[i] << zeros16; + qDomainDiff = zeros16 - dfa_clean_q_domain_diff; + tmp16no2 = ptrDfaClean[i] >> -qDomainDiff; } else { - tmp16no1 = WEBRTC_SPL_SHIFT_W16(aecm->nearFilt[i], - aecm->dfaCleanQDomain - - aecm->dfaCleanQDomainOld); + tmp16no1 = dfa_clean_q_domain_diff < 0 + ? aecm->nearFilt[i] >> -dfa_clean_q_domain_diff + : aecm->nearFilt[i] << dfa_clean_q_domain_diff; qDomainDiff = 0; tmp16no2 = ptrDfaClean[i]; } @@ -1022,7 +1023,8 @@ int WebRtcAecm_ProcessBlock(AecmCore_t* aecm, if ((tmp16no2) & (-qDomainDiff > zeros16)) { aecm->nearFilt[i] = WEBRTC_SPL_WORD16_MAX; } else { - aecm->nearFilt[i] = WEBRTC_SPL_SHIFT_W16(tmp16no2, -qDomainDiff); + aecm->nearFilt[i] = qDomainDiff < 0 ? tmp16no2 << -qDomainDiff + : tmp16no2 >> qDomainDiff; } // Wiener filter coefficients, resulting hnl in Q14 diff --git a/modules/audio_processing/audio_buffer.cc b/modules/audio_processing/audio_buffer.cc index b0f1eb6c..35e1eb7c 100644 --- a/modules/audio_processing/audio_buffer.cc +++ b/modules/audio_processing/audio_buffer.cc @@ -294,11 +294,16 @@ int16_t* AudioBuffer::data(int channel) { return const_cast<int16_t*>(t->data(channel)); } -float* AudioBuffer::data_f(int channel) { +const float* AudioBuffer::data_f(int channel) const { assert(channel >= 0 && channel < num_proc_channels_); return channels_->fbuf()->channel(channel); } +float* AudioBuffer::data_f(int channel) { + const AudioBuffer* t = this; + return const_cast<float*>(t->data_f(channel)); +} + const int16_t* AudioBuffer::low_pass_split_data(int channel) const { assert(channel >= 0 && channel < num_proc_channels_); return split_channels_.get() ? split_channels_->low_channel(channel) @@ -310,12 +315,17 @@ int16_t* AudioBuffer::low_pass_split_data(int channel) { return const_cast<int16_t*>(t->low_pass_split_data(channel)); } -float* AudioBuffer::low_pass_split_data_f(int channel) { +const float* AudioBuffer::low_pass_split_data_f(int channel) const { assert(channel >= 0 && channel < num_proc_channels_); return split_channels_.get() ? split_channels_->low_channel_f(channel) : data_f(channel); } +float* AudioBuffer::low_pass_split_data_f(int channel) { + const AudioBuffer* t = this; + return const_cast<float*>(t->low_pass_split_data_f(channel)); +} + const int16_t* AudioBuffer::high_pass_split_data(int channel) const { assert(channel >= 0 && channel < num_proc_channels_); return split_channels_.get() ? split_channels_->high_channel(channel) : NULL; @@ -326,12 +336,17 @@ int16_t* AudioBuffer::high_pass_split_data(int channel) { return const_cast<int16_t*>(t->high_pass_split_data(channel)); } -float* AudioBuffer::high_pass_split_data_f(int channel) { +const float* AudioBuffer::high_pass_split_data_f(int channel) const { assert(channel >= 0 && channel < num_proc_channels_); return split_channels_.get() ? split_channels_->high_channel_f(channel) : NULL; } +float* AudioBuffer::high_pass_split_data_f(int channel) { + const AudioBuffer* t = this; + return const_cast<float*>(t->high_pass_split_data_f(channel)); +} + const int16_t* AudioBuffer::mixed_data(int channel) const { assert(channel >= 0 && channel < num_mixed_channels_); diff --git a/modules/audio_processing/audio_buffer.h b/modules/audio_processing/audio_buffer.h index 67e4f485..2fab814a 100644 --- a/modules/audio_processing/audio_buffer.h +++ b/modules/audio_processing/audio_buffer.h @@ -69,8 +69,11 @@ class AudioBuffer { // Float versions of the accessors, with automatic conversion back and forth // as necessary. The range of the numbers are the same as for int16_t. float* data_f(int channel); + const float* data_f(int channel) const; float* low_pass_split_data_f(int channel); + const float* low_pass_split_data_f(int channel) const; float* high_pass_split_data_f(int channel); + const float* high_pass_split_data_f(int channel) const; const float* keyboard_data() const; diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc index de387edb..cc2b5c00 100644 --- a/modules/audio_processing/audio_processing_impl.cc +++ b/modules/audio_processing/audio_processing_impl.cc @@ -303,10 +303,6 @@ void AudioProcessingImpl::SetExtraOptions(const Config& config) { (*it)->SetExtraOptions(config); } -int AudioProcessingImpl::EnableExperimentalNs(bool enable) { - return kNoError; -} - int AudioProcessingImpl::input_sample_rate_hz() const { CriticalSectionScoped crit_scoped(crit_); return fwd_in_format_.rate(); diff --git a/modules/audio_processing/audio_processing_impl.h b/modules/audio_processing/audio_processing_impl.h index d34f305a..9753423d 100644 --- a/modules/audio_processing/audio_processing_impl.h +++ b/modules/audio_processing/audio_processing_impl.h @@ -92,10 +92,6 @@ class AudioProcessingImpl : public AudioProcessing { ChannelLayout output_layout, ChannelLayout reverse_layout) OVERRIDE; virtual void SetExtraOptions(const Config& config) OVERRIDE; - virtual int EnableExperimentalNs(bool enable) OVERRIDE; - virtual bool experimental_ns_enabled() const OVERRIDE { - return false; - } virtual int set_sample_rate_hz(int rate) OVERRIDE; virtual int input_sample_rate_hz() const OVERRIDE; virtual int sample_rate_hz() const OVERRIDE; diff --git a/modules/audio_processing/echo_cancellation_impl.cc b/modules/audio_processing/echo_cancellation_impl.cc index e770f9fe..47b4f181 100644 --- a/modules/audio_processing/echo_cancellation_impl.cc +++ b/modules/audio_processing/echo_cancellation_impl.cc @@ -89,7 +89,7 @@ int EchoCancellationImpl::ProcessRenderAudio(const AudioBuffer* audio) { Handle* my_handle = static_cast<Handle*>(handle(handle_index)); err = WebRtcAec_BufferFarend( my_handle, - audio->low_pass_split_data(j), + audio->low_pass_split_data_f(j), static_cast<int16_t>(audio->samples_per_split_channel())); if (err != apm_->kNoError) { diff --git a/modules/audio_processing/include/audio_processing.h b/modules/audio_processing/include/audio_processing.h index 1e494d18..6f4cc9e7 100644 --- a/modules/audio_processing/include/audio_processing.h +++ b/modules/audio_processing/include/audio_processing.h @@ -209,8 +209,8 @@ class AudioProcessing { // ensures the options are applied immediately. virtual void SetExtraOptions(const Config& config) = 0; - virtual int EnableExperimentalNs(bool enable) = 0; - virtual bool experimental_ns_enabled() const = 0; + virtual int EnableExperimentalNs(bool enable) { return kNoError; } + virtual bool experimental_ns_enabled() const { return false; } // DEPRECATED. // TODO(ajm): Remove after Chromium has upgraded to using Initialize(). diff --git a/modules/audio_processing/include/mock_audio_processing.h b/modules/audio_processing/include/mock_audio_processing.h index c1ac23ad..8258bb6c 100644 --- a/modules/audio_processing/include/mock_audio_processing.h +++ b/modules/audio_processing/include/mock_audio_processing.h @@ -186,10 +186,6 @@ class MockAudioProcessing : public AudioProcessing { ChannelLayout reverse_layout)); MOCK_METHOD1(SetExtraOptions, void(const Config& config)); - MOCK_METHOD1(EnableExperimentalNs, - int(bool enable)); - MOCK_CONST_METHOD0(experimental_ns_enabled, - bool()); MOCK_METHOD1(set_sample_rate_hz, int(int rate)); MOCK_CONST_METHOD0(input_sample_rate_hz, diff --git a/modules/audio_processing/utility/delay_estimator.c b/modules/audio_processing/utility/delay_estimator.c index 3b204326..60c7a042 100644 --- a/modules/audio_processing/utility/delay_estimator.c +++ b/modules/audio_processing/utility/delay_estimator.c @@ -98,6 +98,7 @@ static void UpdateRobustValidationStatistics(BinaryDelayEstimator* self, kMaxHitsWhenPossiblyNonCausal : kMaxHitsWhenPossiblyCausal; int i = 0; + assert(self->history_size == self->farend->history_size); // Reset |candidate_hits| if we have a new candidate. if (candidate_delay != self->last_candidate_delay) { self->candidate_hits = 0; @@ -130,7 +131,7 @@ static void UpdateRobustValidationStatistics(BinaryDelayEstimator* self, // 4. All other bins are decreased with |valley_depth|. // TODO(bjornv): Investigate how to make this loop more efficient. Split up // the loop? Remove parts that doesn't add too much. - for (i = 0; i < self->farend->history_size; ++i) { + for (i = 0; i < self->history_size; ++i) { int is_in_last_set = (i >= self->last_delay - 2) && (i <= self->last_delay + 1) && (i != candidate_delay); int is_in_candidate_set = (i >= candidate_delay - 2) && @@ -277,25 +278,45 @@ BinaryDelayEstimatorFarend* WebRtc_CreateBinaryDelayEstimatorFarend( // Sanity conditions fulfilled. self = malloc(sizeof(BinaryDelayEstimatorFarend)); } - if (self != NULL) { - int malloc_fail = 0; - - self->history_size = history_size; - - // Allocate memory for history buffers. - self->binary_far_history = malloc(history_size * sizeof(uint32_t)); - malloc_fail |= (self->binary_far_history == NULL); + if (self == NULL) { + return NULL; + } - self->far_bit_counts = malloc(history_size * sizeof(int)); - malloc_fail |= (self->far_bit_counts == NULL); + self->history_size = 0; + self->binary_far_history = NULL; + self->far_bit_counts = NULL; + if (WebRtc_AllocateFarendBufferMemory(self, history_size) == 0) { + WebRtc_FreeBinaryDelayEstimatorFarend(self); + self = NULL; + } + return self; +} - if (malloc_fail) { - WebRtc_FreeBinaryDelayEstimatorFarend(self); - self = NULL; - } +int WebRtc_AllocateFarendBufferMemory(BinaryDelayEstimatorFarend* self, + int history_size) { + assert(self != NULL); + // (Re-)Allocate memory for history buffers. + self->binary_far_history = + realloc(self->binary_far_history, + history_size * sizeof(*self->binary_far_history)); + self->far_bit_counts = realloc(self->far_bit_counts, + history_size * sizeof(*self->far_bit_counts)); + if ((self->binary_far_history == NULL) || (self->far_bit_counts == NULL)) { + history_size = 0; + } + // Fill with zeros if we have expanded the buffers. + if (history_size > self->history_size) { + int size_diff = history_size - self->history_size; + memset(&self->binary_far_history[self->history_size], + 0, + sizeof(*self->binary_far_history) * size_diff); + memset(&self->far_bit_counts[self->history_size], + 0, + sizeof(*self->far_bit_counts) * size_diff); } + self->history_size = history_size; - return self; + return self->history_size; } void WebRtc_InitBinaryDelayEstimatorFarend(BinaryDelayEstimatorFarend* self) { @@ -385,51 +406,84 @@ BinaryDelayEstimator* WebRtc_CreateBinaryDelayEstimator( // Sanity conditions fulfilled. self = malloc(sizeof(BinaryDelayEstimator)); } + if (self == NULL) { + return NULL; + } - if (self != NULL) { - int malloc_fail = 0; - - self->farend = farend; - self->near_history_size = max_lookahead + 1; - self->robust_validation_enabled = 0; // Disabled by default. - self->allowed_offset = 0; - - self->lookahead = max_lookahead; - - // Allocate memory for spectrum buffers. The extra array element in - // |mean_bit_counts| and |histogram| is a dummy element only used while - // |last_delay| == -2, i.e., before we have a valid estimate. - self->mean_bit_counts = - malloc((farend->history_size + 1) * sizeof(int32_t)); - malloc_fail |= (self->mean_bit_counts == NULL); + self->farend = farend; + self->near_history_size = max_lookahead + 1; + self->history_size = 0; + self->robust_validation_enabled = 0; // Disabled by default. + self->allowed_offset = 0; - self->bit_counts = malloc(farend->history_size * sizeof(int32_t)); - malloc_fail |= (self->bit_counts == NULL); + self->lookahead = max_lookahead; - // Allocate memory for history buffers. - self->binary_near_history = malloc((max_lookahead + 1) * sizeof(uint32_t)); - malloc_fail |= (self->binary_near_history == NULL); + // Allocate memory for spectrum and history buffers. + self->mean_bit_counts = NULL; + self->bit_counts = NULL; + self->histogram = NULL; + self->binary_near_history = + malloc((max_lookahead + 1) * sizeof(*self->binary_near_history)); + if (self->binary_near_history == NULL || + WebRtc_AllocateHistoryBufferMemory(self, farend->history_size) == 0) { + WebRtc_FreeBinaryDelayEstimator(self); + self = NULL; + } - self->histogram = malloc((farend->history_size + 1) * sizeof(float)); - malloc_fail |= (self->histogram == NULL); + return self; +} - if (malloc_fail) { - WebRtc_FreeBinaryDelayEstimator(self); - self = NULL; - } +int WebRtc_AllocateHistoryBufferMemory(BinaryDelayEstimator* self, + int history_size) { + BinaryDelayEstimatorFarend* far = self->farend; + // (Re-)Allocate memory for spectrum and history buffers. + if (history_size != far->history_size) { + // Only update far-end buffers if we need. + history_size = WebRtc_AllocateFarendBufferMemory(far, history_size); + } + // The extra array element in |mean_bit_counts| and |histogram| is a dummy + // element only used while |last_delay| == -2, i.e., before we have a valid + // estimate. + self->mean_bit_counts = + realloc(self->mean_bit_counts, + (history_size + 1) * sizeof(*self->mean_bit_counts)); + self->bit_counts = + realloc(self->bit_counts, history_size * sizeof(*self->bit_counts)); + self->histogram = + realloc(self->histogram, (history_size + 1) * sizeof(*self->histogram)); + + if ((self->mean_bit_counts == NULL) || + (self->bit_counts == NULL) || + (self->histogram == NULL)) { + history_size = 0; } + // Fill with zeros if we have expanded the buffers. + if (history_size > self->history_size) { + int size_diff = history_size - self->history_size; + memset(&self->mean_bit_counts[self->history_size], + 0, + sizeof(*self->mean_bit_counts) * size_diff); + memset(&self->bit_counts[self->history_size], + 0, + sizeof(*self->bit_counts) * size_diff); + memset(&self->histogram[self->history_size], + 0, + sizeof(*self->histogram) * size_diff); + } + self->history_size = history_size; - return self; + return self->history_size; } void WebRtc_InitBinaryDelayEstimator(BinaryDelayEstimator* self) { int i = 0; assert(self != NULL); - memset(self->bit_counts, 0, sizeof(int32_t) * self->farend->history_size); - memset(self->binary_near_history, 0, + memset(self->bit_counts, 0, sizeof(int32_t) * self->history_size); + memset(self->binary_near_history, + 0, sizeof(uint32_t) * self->near_history_size); - for (i = 0; i <= self->farend->history_size; ++i) { + for (i = 0; i <= self->history_size; ++i) { self->mean_bit_counts[i] = (20 << 9); // 20 in Q9. self->histogram[i] = 0.f; } @@ -440,7 +494,7 @@ void WebRtc_InitBinaryDelayEstimator(BinaryDelayEstimator* self) { self->last_delay = -2; self->last_candidate_delay = -2; - self->compare_delay = self->farend->history_size; + self->compare_delay = self->history_size; self->candidate_hits = 0; self->last_delay_histogram = 0.f; } @@ -471,6 +525,10 @@ int WebRtc_ProcessBinarySpectrum(BinaryDelayEstimator* self, int32_t valley_depth = 0; assert(self != NULL); + if (self->farend->history_size != self->history_size) { + // Non matching history sizes. + return -1; + } if (self->near_history_size > 1) { // If we apply lookahead, shift near-end binary spectrum history. Insert // current |binary_near_spectrum| and pull out the delayed one. @@ -482,10 +540,10 @@ int WebRtc_ProcessBinarySpectrum(BinaryDelayEstimator* self, // Compare with delayed spectra and store the |bit_counts| for each delay. BitCountComparison(binary_near_spectrum, self->farend->binary_far_history, - self->farend->history_size, self->bit_counts); + self->history_size, self->bit_counts); // Update |mean_bit_counts|, which is the smoothed version of |bit_counts|. - for (i = 0; i < self->farend->history_size; i++) { + for (i = 0; i < self->history_size; i++) { // |bit_counts| is constrained to [0, 32], meaning we can smooth with a // factor up to 2^26. We use Q9. int32_t bit_count = (self->bit_counts[i] << 9); // Q9. @@ -503,7 +561,7 @@ int WebRtc_ProcessBinarySpectrum(BinaryDelayEstimator* self, // Find |candidate_delay|, |value_best_candidate| and |value_worst_candidate| // of |mean_bit_counts|. - for (i = 0; i < self->farend->history_size; i++) { + for (i = 0; i < self->history_size; i++) { if (self->mean_bit_counts[i] < value_best_candidate) { value_best_candidate = self->mean_bit_counts[i]; candidate_delay = i; diff --git a/modules/audio_processing/utility/delay_estimator.h b/modules/audio_processing/utility/delay_estimator.h index 3d5ffce2..65c3f034 100644 --- a/modules/audio_processing/utility/delay_estimator.h +++ b/modules/audio_processing/utility/delay_estimator.h @@ -36,6 +36,7 @@ typedef struct { // Binary history variables. uint32_t* binary_near_history; int near_history_size; + int history_size; // Delay estimation variables. int32_t minimum_probability; @@ -85,6 +86,19 @@ void WebRtc_FreeBinaryDelayEstimatorFarend(BinaryDelayEstimatorFarend* self); BinaryDelayEstimatorFarend* WebRtc_CreateBinaryDelayEstimatorFarend( int history_size); +// Re-allocates the buffers. +// +// Inputs: +// - self : Pointer to the binary estimation far-end instance +// which is the return value of +// WebRtc_CreateBinaryDelayEstimatorFarend(). +// - history_size : Size of the far-end binary spectrum history. +// +// Return value: +// - history_size : The history size allocated. +int WebRtc_AllocateFarendBufferMemory(BinaryDelayEstimatorFarend* self, + int history_size); + // Initializes the delay estimation far-end instance created with // WebRtc_CreateBinaryDelayEstimatorFarend(...). // @@ -141,6 +155,20 @@ void WebRtc_FreeBinaryDelayEstimator(BinaryDelayEstimator* self); BinaryDelayEstimator* WebRtc_CreateBinaryDelayEstimator( BinaryDelayEstimatorFarend* farend, int max_lookahead); +// Re-allocates |history_size| dependent buffers. The far-end buffers will be +// updated at the same time if needed. +// +// Input: +// - self : Pointer to the binary estimation instance which is +// the return value of +// WebRtc_CreateBinaryDelayEstimator(). +// - history_size : Size of the history buffers. +// +// Return value: +// - history_size : The history size allocated. +int WebRtc_AllocateHistoryBufferMemory(BinaryDelayEstimator* self, + int history_size); + // Initializes the delay estimation instance created with // WebRtc_CreateBinaryDelayEstimator(...). // @@ -220,5 +248,4 @@ void WebRtc_MeanEstimatorFix(int32_t new_value, int factor, int32_t* mean_value); - #endif // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_H_ diff --git a/modules/audio_processing/utility/delay_estimator_unittest.cc b/modules/audio_processing/utility/delay_estimator_unittest.cc index ca0901d6..4ebe0e61 100644 --- a/modules/audio_processing/utility/delay_estimator_unittest.cc +++ b/modules/audio_processing/utility/delay_estimator_unittest.cc @@ -23,9 +23,13 @@ enum { kSpectrumSize = 65 }; // Delay history sizes. enum { kMaxDelay = 100 }; enum { kLookahead = 10 }; +enum { kHistorySize = kMaxDelay + kLookahead }; // Length of binary spectrum sequence. enum { kSequenceLength = 400 }; +const int kDifferentHistorySize = 3; +const int kDifferentLookahead = 1; + const int kEnable[] = { 0, 1 }; const size_t kSizeEnable = sizeof(kEnable) / sizeof(*kEnable); @@ -56,7 +60,7 @@ class DelayEstimatorTest : public ::testing::Test { float near_f_[kSpectrumSize]; uint16_t far_u16_[kSpectrumSize]; uint16_t near_u16_[kSpectrumSize]; - uint32_t binary_spectrum_[kSequenceLength + kMaxDelay + kLookahead]; + uint32_t binary_spectrum_[kSequenceLength + kHistorySize]; }; DelayEstimatorTest::DelayEstimatorTest() @@ -76,21 +80,20 @@ DelayEstimatorTest::DelayEstimatorTest() // |kSequenceLength| has to be long enough for the delay estimation to leave // the initialized state. binary_spectrum_[0] = 1; - for (int i = 1; i < (kSequenceLength + kMaxDelay + kLookahead); i++) { + for (int i = 1; i < (kSequenceLength + kHistorySize); i++) { binary_spectrum_[i] = 3 * binary_spectrum_[i - 1]; } } void DelayEstimatorTest::SetUp() { farend_handle_ = WebRtc_CreateDelayEstimatorFarend(kSpectrumSize, - kMaxDelay + kLookahead); + kHistorySize); ASSERT_TRUE(farend_handle_ != NULL); farend_self_ = reinterpret_cast<DelayEstimatorFarend*>(farend_handle_); handle_ = WebRtc_CreateDelayEstimator(farend_handle_, kLookahead); ASSERT_TRUE(handle_ != NULL); self_ = reinterpret_cast<DelayEstimator*>(handle_); - binary_farend_ = WebRtc_CreateBinaryDelayEstimatorFarend(kMaxDelay + - kLookahead); + binary_farend_ = WebRtc_CreateBinaryDelayEstimatorFarend(kHistorySize); ASSERT_TRUE(binary_farend_ != NULL); binary_ = WebRtc_CreateBinaryDelayEstimator(binary_farend_, kLookahead); ASSERT_TRUE(binary_ != NULL); @@ -226,7 +229,7 @@ TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfWrapper) { // Make sure we have a non-NULL value at start, so we can detect NULL after // create failure. void* handle = farend_handle_; - handle = WebRtc_CreateDelayEstimatorFarend(33, kMaxDelay + kLookahead); + handle = WebRtc_CreateDelayEstimatorFarend(33, kHistorySize); EXPECT_TRUE(handle == NULL); handle = WebRtc_CreateDelayEstimatorFarend(kSpectrumSize, 1); EXPECT_TRUE(handle == NULL); @@ -266,6 +269,28 @@ TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfWrapper) { EXPECT_EQ(-1, WebRtc_AddFarSpectrumFix(farend_handle_, far_u16_, spectrum_size_, 16)); + // WebRtc_set_history_size() should return -1 if: + // 1) |handle| is a NULL. + // 2) |history_size| <= 1. + EXPECT_EQ(-1, WebRtc_set_history_size(NULL, 1)); + EXPECT_EQ(-1, WebRtc_set_history_size(handle_, 1)); + // WebRtc_history_size() should return -1 if: + // 1) NULL pointer input. + EXPECT_EQ(-1, WebRtc_history_size(NULL)); + // 2) there is a mismatch between history size. + void* tmp_handle = WebRtc_CreateDelayEstimator(farend_handle_, kHistorySize); + EXPECT_EQ(0, WebRtc_InitDelayEstimator(tmp_handle)); + EXPECT_EQ(kDifferentHistorySize, + WebRtc_set_history_size(tmp_handle, kDifferentHistorySize)); + EXPECT_EQ(kDifferentHistorySize, WebRtc_history_size(tmp_handle)); + EXPECT_EQ(kHistorySize, WebRtc_set_history_size(handle_, kHistorySize)); + EXPECT_EQ(-1, WebRtc_history_size(tmp_handle)); + + // WebRtc_set_lookahead() should return -1 if we try a value outside the + /// buffer. + EXPECT_EQ(-1, WebRtc_set_lookahead(handle_, kLookahead + 1)); + EXPECT_EQ(-1, WebRtc_set_lookahead(handle_, -1)); + // WebRtc_set_allowed_offset() should return -1 if we have: // 1) NULL pointer as |handle|. // 2) |allowed_offset| < 0. @@ -289,6 +314,8 @@ TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfWrapper) { // 1) NULL pointer as |handle|. // 2) NULL pointer as near-end spectrum. // 3) Incorrect spectrum size. + // 4) Non matching history sizes if multiple delay estimators using the same + // far-end reference. EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFloat(NULL, near_f_, spectrum_size_)); // Use |handle_| which is properly created at SetUp(). @@ -296,12 +323,18 @@ TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfWrapper) { spectrum_size_)); EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFloat(handle_, near_f_, spectrum_size_ + 1)); + // |tmp_handle| is already in a non-matching state. + EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFloat(tmp_handle, + near_f_, + spectrum_size_)); // WebRtc_DelayEstimatorProcessFix() should return -1 if we have: // 1) NULL pointer as |handle|. - // 3) NULL pointer as near-end spectrum. - // 4) Incorrect spectrum size. - // 6) Too high precision in near-end spectrum (Q-domain > 15). + // 2) NULL pointer as near-end spectrum. + // 3) Incorrect spectrum size. + // 4) Too high precision in near-end spectrum (Q-domain > 15). + // 5) Non matching history sizes if multiple delay estimators using the same + // far-end reference. EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFix(NULL, near_u16_, spectrum_size_, 0)); EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFix(handle_, NULL, spectrum_size_, @@ -310,6 +343,12 @@ TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfWrapper) { spectrum_size_ + 1, 0)); EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFix(handle_, near_u16_, spectrum_size_, 16)); + // |tmp_handle| is already in a non-matching state. + EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFix(tmp_handle, + near_u16_, + spectrum_size_, + 0)); + WebRtc_FreeDelayEstimator(tmp_handle); // WebRtc_last_delay() should return -1 if we have a NULL pointer as |handle|. EXPECT_EQ(-1, WebRtc_last_delay(NULL)); @@ -344,14 +383,23 @@ TEST_F(DelayEstimatorTest, VerifyEnableRobustValidation) { TEST_F(DelayEstimatorTest, InitializedSpectrumAfterProcess) { // In this test we verify that the mean spectra are initialized after first - // time we call WebRtc_AddFarSpectrum() and Process() respectively. + // time we call WebRtc_AddFarSpectrum() and Process() respectively. The test + // also verifies the state is not left for zero spectra. + const float kZerosFloat[kSpectrumSize] = { 0.0 }; + const uint16_t kZerosU16[kSpectrumSize] = { 0 }; // For floating point operations, process one frame and verify initialization // flag. Init(); + EXPECT_EQ(0, WebRtc_AddFarSpectrumFloat(farend_handle_, kZerosFloat, + spectrum_size_)); + EXPECT_EQ(0, farend_self_->far_spectrum_initialized); EXPECT_EQ(0, WebRtc_AddFarSpectrumFloat(farend_handle_, far_f_, spectrum_size_)); EXPECT_EQ(1, farend_self_->far_spectrum_initialized); + EXPECT_EQ(-2, WebRtc_DelayEstimatorProcessFloat(handle_, kZerosFloat, + spectrum_size_)); + EXPECT_EQ(0, self_->near_spectrum_initialized); EXPECT_EQ(-2, WebRtc_DelayEstimatorProcessFloat(handle_, near_f_, spectrum_size_)); EXPECT_EQ(1, self_->near_spectrum_initialized); @@ -359,9 +407,15 @@ TEST_F(DelayEstimatorTest, InitializedSpectrumAfterProcess) { // For fixed point operations, process one frame and verify initialization // flag. Init(); + EXPECT_EQ(0, WebRtc_AddFarSpectrumFix(farend_handle_, kZerosU16, + spectrum_size_, 0)); + EXPECT_EQ(0, farend_self_->far_spectrum_initialized); EXPECT_EQ(0, WebRtc_AddFarSpectrumFix(farend_handle_, far_u16_, spectrum_size_, 0)); EXPECT_EQ(1, farend_self_->far_spectrum_initialized); + EXPECT_EQ(-2, WebRtc_DelayEstimatorProcessFix(handle_, kZerosU16, + spectrum_size_, 0)); + EXPECT_EQ(0, self_->near_spectrum_initialized); EXPECT_EQ(-2, WebRtc_DelayEstimatorProcessFix(handle_, near_u16_, spectrum_size_, 0)); EXPECT_EQ(1, self_->near_spectrum_initialized); @@ -536,6 +590,32 @@ TEST_F(DelayEstimatorTest, VerifyLookaheadAtCreate) { WebRtc_FreeDelayEstimatorFarend(farend_handle); } +TEST_F(DelayEstimatorTest, VerifyLookaheadIsSetAndKeptAfterInit) { + EXPECT_EQ(kLookahead, WebRtc_lookahead(handle_)); + EXPECT_EQ(kDifferentLookahead, + WebRtc_set_lookahead(handle_, kDifferentLookahead)); + EXPECT_EQ(kDifferentLookahead, WebRtc_lookahead(handle_)); + EXPECT_EQ(0, WebRtc_InitDelayEstimatorFarend(farend_handle_)); + EXPECT_EQ(kDifferentLookahead, WebRtc_lookahead(handle_)); + EXPECT_EQ(0, WebRtc_InitDelayEstimator(handle_)); + EXPECT_EQ(kDifferentLookahead, WebRtc_lookahead(handle_)); +} + +TEST_F(DelayEstimatorTest, VerifyHistorySizeAtCreate) { + EXPECT_EQ(kHistorySize, WebRtc_history_size(handle_)); +} + +TEST_F(DelayEstimatorTest, VerifyHistorySizeIsSetAndKeptAfterInit) { + EXPECT_EQ(kHistorySize, WebRtc_history_size(handle_)); + EXPECT_EQ(kDifferentHistorySize, + WebRtc_set_history_size(handle_, kDifferentHistorySize)); + EXPECT_EQ(kDifferentHistorySize, WebRtc_history_size(handle_)); + EXPECT_EQ(0, WebRtc_InitDelayEstimator(handle_)); + EXPECT_EQ(kDifferentHistorySize, WebRtc_history_size(handle_)); + EXPECT_EQ(0, WebRtc_InitDelayEstimatorFarend(farend_handle_)); + EXPECT_EQ(kDifferentHistorySize, WebRtc_history_size(handle_)); +} + // TODO(bjornv): Add tests for SoftReset...(...). } // namespace diff --git a/modules/audio_processing/utility/delay_estimator_wrapper.c b/modules/audio_processing/utility/delay_estimator_wrapper.c index 6ec894e6..270588f3 100644 --- a/modules/audio_processing/utility/delay_estimator_wrapper.c +++ b/modules/audio_processing/utility/delay_estimator_wrapper.c @@ -58,7 +58,7 @@ static void MeanEstimatorFloat(float new_value, // Return: // - out : Binary spectrum. // -static uint32_t BinarySpectrumFix(uint16_t* spectrum, +static uint32_t BinarySpectrumFix(const uint16_t* spectrum, SpectrumType* threshold_spectrum, int q_domain, int* threshold_initialized) { @@ -93,7 +93,7 @@ static uint32_t BinarySpectrumFix(uint16_t* spectrum, return out; } -static uint32_t BinarySpectrumFloat(float* spectrum, +static uint32_t BinarySpectrumFloat(const float* spectrum, SpectrumType* threshold_spectrum, int* threshold_initialized) { int i = kBandFirst; @@ -147,7 +147,7 @@ void* WebRtc_CreateDelayEstimatorFarend(int spectrum_size, int history_size) { COMPILE_ASSERT(kBandLast - kBandFirst < 32); if (spectrum_size >= kBandLast) { - self = malloc(sizeof(DelayEstimator)); + self = malloc(sizeof(DelayEstimatorFarend)); } if (self != NULL) { @@ -197,8 +197,10 @@ void WebRtc_SoftResetDelayEstimatorFarend(void* handle, int delay_shift) { WebRtc_SoftResetBinaryDelayEstimatorFarend(self->binary_farend, delay_shift); } -int WebRtc_AddFarSpectrumFix(void* handle, uint16_t* far_spectrum, - int spectrum_size, int far_q) { +int WebRtc_AddFarSpectrumFix(void* handle, + const uint16_t* far_spectrum, + int spectrum_size, + int far_q) { DelayEstimatorFarend* self = (DelayEstimatorFarend*) handle; uint32_t binary_spectrum = 0; @@ -226,7 +228,8 @@ int WebRtc_AddFarSpectrumFix(void* handle, uint16_t* far_spectrum, return 0; } -int WebRtc_AddFarSpectrumFloat(void* handle, float* far_spectrum, +int WebRtc_AddFarSpectrumFloat(void* handle, + const float* far_spectrum, int spectrum_size) { DelayEstimatorFarend* self = (DelayEstimatorFarend*) handle; uint32_t binary_spectrum = 0; @@ -324,6 +327,29 @@ int WebRtc_SoftResetDelayEstimator(void* handle, int delay_shift) { return WebRtc_SoftResetBinaryDelayEstimator(self->binary_handle, delay_shift); } +int WebRtc_set_history_size(void* handle, int history_size) { + DelayEstimator* self = handle; + + if ((self == NULL) || (history_size <= 1)) { + return -1; + } + return WebRtc_AllocateHistoryBufferMemory(self->binary_handle, history_size); +} + +int WebRtc_history_size(const void* handle) { + const DelayEstimator* self = handle; + + if (self == NULL) { + return -1; + } + if (self->binary_handle->farend->history_size != + self->binary_handle->history_size) { + // Non matching history sizes. + return -1; + } + return self->binary_handle->history_size; +} + int WebRtc_set_lookahead(void* handle, int lookahead) { DelayEstimator* self = (DelayEstimator*) handle; assert(self != NULL); @@ -386,7 +412,7 @@ int WebRtc_is_robust_validation_enabled(const void* handle) { } int WebRtc_DelayEstimatorProcessFix(void* handle, - uint16_t* near_spectrum, + const uint16_t* near_spectrum, int spectrum_size, int near_q) { DelayEstimator* self = (DelayEstimator*) handle; @@ -418,7 +444,7 @@ int WebRtc_DelayEstimatorProcessFix(void* handle, } int WebRtc_DelayEstimatorProcessFloat(void* handle, - float* near_spectrum, + const float* near_spectrum, int spectrum_size) { DelayEstimator* self = (DelayEstimator*) handle; uint32_t binary_spectrum = 0; diff --git a/modules/audio_processing/utility/delay_estimator_wrapper.h b/modules/audio_processing/utility/delay_estimator_wrapper.h index 13e86bdd..fdadebeb 100644 --- a/modules/audio_processing/utility/delay_estimator_wrapper.h +++ b/modules/audio_processing/utility/delay_estimator_wrapper.h @@ -17,9 +17,6 @@ #include "webrtc/typedefs.h" // Releases the memory allocated by WebRtc_CreateDelayEstimatorFarend(...) -// Input: -// - handle : Pointer to the delay estimation far-end instance. -// void WebRtc_FreeDelayEstimatorFarend(void* handle); // Allocates the memory needed by the far-end part of the delay estimation. The @@ -27,36 +24,28 @@ void WebRtc_FreeDelayEstimatorFarend(void* handle); // WebRtc_InitDelayEstimatorFarend(...). // // Inputs: -// - spectrum_size : Size of the spectrum used both in far-end and +// - spectrum_size : Size of the spectrum used both in far-end and // near-end. Used to allocate memory for spectrum // specific buffers. -// - history_size : The far-end history buffer size. Note that the maximum -// delay which can be estimated is controlled together -// with |lookahead| through -// WebRtc_CreateDelayEstimator(). +// - history_size : The far-end history buffer size. A change in buffer +// size can be forced with WebRtc_set_history_size(). +// Note that the maximum delay which can be estimated is +// determined together with WebRtc_set_lookahead(). // // Return value: -// - void* : Created |handle|. If the memory can't be allocated or +// - void* : Created |handle|. If the memory can't be allocated or // if any of the input parameters are invalid NULL is // returned. -// void* WebRtc_CreateDelayEstimatorFarend(int spectrum_size, int history_size); // Initializes the far-end part of the delay estimation instance returned by // WebRtc_CreateDelayEstimatorFarend(...) -// Input: -// - handle : Pointer to the delay estimation far-end instance. -// -// Output: -// - handle : Initialized instance. -// int WebRtc_InitDelayEstimatorFarend(void* handle); // Soft resets the far-end part of the delay estimation instance returned by // WebRtc_CreateDelayEstimatorFarend(...). // Input: // - delay_shift : The amount of blocks to shift history buffers. -// void WebRtc_SoftResetDelayEstimatorFarend(void* handle, int delay_shift); // Adds the far-end spectrum to the far-end history buffer. This spectrum is @@ -64,7 +53,6 @@ void WebRtc_SoftResetDelayEstimatorFarend(void* handle, int delay_shift); // WebRtc_ProcessSpectrum(). // // Inputs: -// - handle : Pointer to the delay estimation far-end instance. // - far_spectrum : Far-end spectrum. // - spectrum_size : The size of the data arrays (same for both far- and // near-end). @@ -73,17 +61,17 @@ void WebRtc_SoftResetDelayEstimatorFarend(void* handle, int delay_shift); // Output: // - handle : Updated far-end instance. // -int WebRtc_AddFarSpectrumFix(void* handle, uint16_t* far_spectrum, - int spectrum_size, int far_q); +int WebRtc_AddFarSpectrumFix(void* handle, + const uint16_t* far_spectrum, + int spectrum_size, + int far_q); // See WebRtc_AddFarSpectrumFix() for description. -int WebRtc_AddFarSpectrumFloat(void* handle, float* far_spectrum, +int WebRtc_AddFarSpectrumFloat(void* handle, + const float* far_spectrum, int spectrum_size); // Releases the memory allocated by WebRtc_CreateDelayEstimator(...) -// Input: -// - handle : Pointer to the delay estimation instance. -// void WebRtc_FreeDelayEstimator(void* handle); // Allocates the memory needed by the delay estimation. The memory needs to be @@ -117,24 +105,17 @@ void WebRtc_FreeDelayEstimator(void* handle); // // Note that the effective range of delay estimates is // [-|lookahead|,... ,|history_size|-|lookahead|) -// where |history_size| was set upon creating the far-end -// history buffer size. +// where |history_size| is set through +// WebRtc_set_history_size(). // // Return value: // - void* : Created |handle|. If the memory can't be allocated or // if any of the input parameters are invalid NULL is // returned. -// void* WebRtc_CreateDelayEstimator(void* farend_handle, int max_lookahead); // Initializes the delay estimation instance returned by // WebRtc_CreateDelayEstimator(...) -// Input: -// - handle : Pointer to the delay estimation instance. -// -// Output: -// - handle : Initialized instance. -// int WebRtc_InitDelayEstimator(void* handle); // Soft resets the delay estimation instance returned by @@ -144,22 +125,44 @@ int WebRtc_InitDelayEstimator(void* handle); // // Return value: // - actual_shifts : The actual number of shifts performed. -// int WebRtc_SoftResetDelayEstimator(void* handle, int delay_shift); +// Sets the effective |history_size| used. Valid values from 2. We simply need +// at least two delays to compare to perform an estimate. If |history_size| is +// changed, buffers are reallocated filling in with zeros if necessary. +// Note that changing the |history_size| affects both buffers in far-end and +// near-end. Hence it is important to change all DelayEstimators that use the +// same reference far-end, to the same |history_size| value. +// Inputs: +// - handle : Pointer to the delay estimation instance. +// - history_size : Effective history size to be used. +// Return value: +// - new_history_size : The new history size used. If the memory was not able +// to be allocated 0 is returned. +int WebRtc_set_history_size(void* handle, int history_size); + +// Returns the history_size currently used. +// Input: +// - handle : Pointer to the delay estimation instance. +int WebRtc_history_size(const void* handle); + // Sets the amount of |lookahead| to use. Valid values are [0, max_lookahead] // where |max_lookahead| was set at create time through // WebRtc_CreateDelayEstimator(...). // // Input: -// - lookahead : The amount of blocks to shift history buffers. +// - handle : Pointer to the delay estimation instance. +// - lookahead : The amount of lookahead to be used. // // Return value: -// - new_lookahead : The actual number of shifts performed. -// +// - new_lookahead : The actual amount of lookahead set, unless |handle| is +// a NULL pointer or |lookahead| is invalid, for which an +// error is returned. int WebRtc_set_lookahead(void* handle, int lookahead); // Returns the amount of lookahead we currently use. +// Input: +// - handle : Pointer to the delay estimation instance. int WebRtc_lookahead(void* handle); // Sets the |allowed_offset| used in the robust validation scheme. If the @@ -178,8 +181,6 @@ int WebRtc_set_allowed_offset(void* handle, int allowed_offset); // Returns the |allowed_offset| in number of partitions. int WebRtc_get_allowed_offset(const void* handle); -// TODO(bjornv): Implement this functionality. Currently, enabling it has no -// impact, hence this is an empty API. // Enables/Disables a robust validation functionality in the delay estimation. // This is by default set to disabled at create time. The state is preserved // over a reset. @@ -209,15 +210,14 @@ int WebRtc_is_robust_validation_enabled(const void* handle); // - delay : >= 0 - Calculated delay value. // -1 - Error. // -2 - Insufficient data for estimation. -// int WebRtc_DelayEstimatorProcessFix(void* handle, - uint16_t* near_spectrum, + const uint16_t* near_spectrum, int spectrum_size, int near_q); // See WebRtc_DelayEstimatorProcessFix() for description. int WebRtc_DelayEstimatorProcessFloat(void* handle, - float* near_spectrum, + const float* near_spectrum, int spectrum_size); // Returns the last calculated delay updated by the function @@ -230,7 +230,6 @@ int WebRtc_DelayEstimatorProcessFloat(void* handle, // - delay : >= 0 - Last calculated delay value. // -1 - Error. // -2 - Insufficient data for estimation. -// int WebRtc_last_delay(void* handle); // Returns the estimation quality/probability of the last calculated delay diff --git a/modules/desktop_capture/BUILD.gn b/modules/desktop_capture/BUILD.gn index bb219e2b..be9658f8 100644 --- a/modules/desktop_capture/BUILD.gn +++ b/modules/desktop_capture/BUILD.gn @@ -37,10 +37,12 @@ source_set("desktop_capture") { "mac/desktop_configuration.mm", "mac/desktop_configuration_monitor.h", "mac/desktop_configuration_monitor.cc", - "mac/osx_version.h", - "mac/osx_version.cc", + "mac/full_screen_chrome_window_detector.cc", + "mac/full_screen_chrome_window_detector.h", "mac/scoped_pixel_buffer_object.cc", "mac/scoped_pixel_buffer_object.h", + "mac/window_list_utils.cc", + "mac/window_list_utils.h", "mouse_cursor.cc", "mouse_cursor.h", "mouse_cursor_monitor.h", diff --git a/modules/desktop_capture/OWNERS b/modules/desktop_capture/OWNERS index 4c0340d6..67d2fa19 100644 --- a/modules/desktop_capture/OWNERS +++ b/modules/desktop_capture/OWNERS @@ -1,4 +1,5 @@ alexeypa@chromium.org +jiayl@webrtc.org sergeyu@chromium.org wez@chromium.org diff --git a/modules/desktop_capture/desktop_capture.gypi b/modules/desktop_capture/desktop_capture.gypi index 6f4a0830..a0195d66 100644 --- a/modules/desktop_capture/desktop_capture.gypi +++ b/modules/desktop_capture/desktop_capture.gypi @@ -38,10 +38,12 @@ "mac/desktop_configuration.mm", "mac/desktop_configuration_monitor.h", "mac/desktop_configuration_monitor.cc", - "mac/osx_version.h", - "mac/osx_version.cc", + "mac/full_screen_chrome_window_detector.cc", + "mac/full_screen_chrome_window_detector.h", "mac/scoped_pixel_buffer_object.cc", "mac/scoped_pixel_buffer_object.h", + "mac/window_list_utils.cc", + "mac/window_list_utils.h", "mouse_cursor.cc", "mouse_cursor.h", "mouse_cursor_monitor.h", diff --git a/modules/desktop_capture/desktop_capture_options.h b/modules/desktop_capture/desktop_capture_options.h index c6aabd45..030cb2b7 100644 --- a/modules/desktop_capture/desktop_capture_options.h +++ b/modules/desktop_capture/desktop_capture_options.h @@ -19,6 +19,7 @@ #if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS) #include "webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h" +#include "webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.h" #endif namespace webrtc { @@ -50,6 +51,14 @@ class DesktopCaptureOptions { void set_configuration_monitor(scoped_refptr<DesktopConfigurationMonitor> m) { configuration_monitor_ = m; } + + FullScreenChromeWindowDetector* full_screen_chrome_window_detector() const { + return full_screen_window_detector_; + } + void set_full_screen_chrome_window_detector( + scoped_refptr<FullScreenChromeWindowDetector> detector) { + full_screen_window_detector_ = detector; + } #endif // Flag indicating that the capturer should use screen change notifications. @@ -82,6 +91,7 @@ class DesktopCaptureOptions { #if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS) scoped_refptr<DesktopConfigurationMonitor> configuration_monitor_; + scoped_refptr<FullScreenChromeWindowDetector> full_screen_window_detector_; #endif #if defined(WEBRTC_WIN) diff --git a/modules/desktop_capture/mac/full_screen_chrome_window_detector.cc b/modules/desktop_capture/mac/full_screen_chrome_window_detector.cc new file mode 100644 index 00000000..23c432f6 --- /dev/null +++ b/modules/desktop_capture/mac/full_screen_chrome_window_detector.cc @@ -0,0 +1,244 @@ +/* + * Copyright (c) 2014 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/desktop_capture/mac/full_screen_chrome_window_detector.h" + +#include <assert.h> +#include <libproc.h> +#include <string> + +#include "webrtc/base/macutils.h" +#include "webrtc/modules/desktop_capture/mac/desktop_configuration.h" +#include "webrtc/modules/desktop_capture/mac/window_list_utils.h" +#include "webrtc/system_wrappers/interface/logging.h" + + +namespace webrtc { + +namespace { + +const int64_t kUpdateIntervalMs = 500; + +// Returns true if the window is minimized. +bool IsWindowMinimized(CGWindowID id) { + CFArrayRef window_id_array = + CFArrayCreate(NULL, reinterpret_cast<const void **>(&id), 1, NULL); + CFArrayRef window_array = + CGWindowListCreateDescriptionFromArray(window_id_array); + bool minimized = false; + + if (window_array && CFArrayGetCount(window_array)) { + CFDictionaryRef window = reinterpret_cast<CFDictionaryRef>( + CFArrayGetValueAtIndex(window_array, 0)); + CFBooleanRef on_screen = reinterpret_cast<CFBooleanRef>( + CFDictionaryGetValue(window, kCGWindowIsOnscreen)); + + minimized = !on_screen; + } + + CFRelease(window_id_array); + CFRelease(window_array); + + return minimized; +} + +// Returns true if the window is occupying a full screen. +bool IsWindowFullScreen(const MacDesktopConfiguration& desktop_config, + CFDictionaryRef window) { + bool fullscreen = false; + + CFDictionaryRef bounds_ref = reinterpret_cast<CFDictionaryRef>( + CFDictionaryGetValue(window, kCGWindowBounds)); + + CGRect bounds; + if (bounds_ref && + CGRectMakeWithDictionaryRepresentation(bounds_ref, &bounds)) { + for (MacDisplayConfigurations::const_iterator it = + desktop_config.displays.begin(); + it != desktop_config.displays.end(); ++it) { + if (it->bounds.equals(DesktopRect::MakeXYWH(bounds.origin.x, + bounds.origin.y, + bounds.size.width, + bounds.size.height))) { + fullscreen = true; + break; + } + } + } + + return fullscreen; +} + +std::string GetWindowTitle(CGWindowID id) { + CFArrayRef window_id_array = + CFArrayCreate(NULL, reinterpret_cast<const void **>(&id), 1, NULL); + CFArrayRef window_array = + CGWindowListCreateDescriptionFromArray(window_id_array); + std::string title; + + if (window_array && CFArrayGetCount(window_array)) { + CFDictionaryRef window = reinterpret_cast<CFDictionaryRef>( + CFArrayGetValueAtIndex(window_array, 0)); + CFStringRef title_ref = reinterpret_cast<CFStringRef>( + CFDictionaryGetValue(window, kCGWindowName)); + + if (title_ref) + rtc::ToUtf8(title_ref, &title); + } + CFRelease(window_id_array); + CFRelease(window_array); + + return title; +} + +int GetWindowOwnerPid(CGWindowID id) { + CFArrayRef window_id_array = + CFArrayCreate(NULL, reinterpret_cast<const void **>(&id), 1, NULL); + CFArrayRef window_array = + CGWindowListCreateDescriptionFromArray(window_id_array); + int pid = 0; + + if (window_array && CFArrayGetCount(window_array)) { + CFDictionaryRef window = reinterpret_cast<CFDictionaryRef>( + CFArrayGetValueAtIndex(window_array, 0)); + CFNumberRef pid_ref = reinterpret_cast<CFNumberRef>( + CFDictionaryGetValue(window, kCGWindowOwnerPID)); + + if (pid_ref) + CFNumberGetValue(pid_ref, kCFNumberIntType, &pid); + } + CFRelease(window_id_array); + CFRelease(window_array); + + return pid; +} + +// Returns the window that is full-screen and has the same title and owner pid +// as the input window. +CGWindowID FindFullScreenWindowWithSamePidAndTitle(CGWindowID id) { + int pid = GetWindowOwnerPid(id); + std::string title = GetWindowTitle(id); + + // Only get on screen, non-desktop windows. + CFArrayRef window_array = CGWindowListCopyWindowInfo( + kCGWindowListOptionOnScreenOnly | kCGWindowListExcludeDesktopElements, + kCGNullWindowID); + if (!window_array) + return kCGNullWindowID; + + CGWindowID full_screen_window = kCGNullWindowID; + + MacDesktopConfiguration desktop_config = MacDesktopConfiguration::GetCurrent( + MacDesktopConfiguration::TopLeftOrigin); + + // Check windows to make sure they have an id, title, and use window layer + // other than 0. + CFIndex count = CFArrayGetCount(window_array); + for (CFIndex i = 0; i < count; ++i) { + CFDictionaryRef window = reinterpret_cast<CFDictionaryRef>( + CFArrayGetValueAtIndex(window_array, i)); + CFStringRef window_title_ref = reinterpret_cast<CFStringRef>( + CFDictionaryGetValue(window, kCGWindowName)); + CFNumberRef window_id_ref = reinterpret_cast<CFNumberRef>( + CFDictionaryGetValue(window, kCGWindowNumber)); + CFNumberRef window_pid_ref = reinterpret_cast<CFNumberRef>( + CFDictionaryGetValue(window, kCGWindowOwnerPID)); + + if (!window_title_ref || !window_id_ref || !window_pid_ref) + continue; + + int window_pid = 0; + CFNumberGetValue(window_pid_ref, kCFNumberIntType, &window_pid); + if (window_pid != pid) + continue; + + std::string window_title; + if (!rtc::ToUtf8(window_title_ref, &window_title) || + window_title != title) { + continue; + } + + CGWindowID window_id; + CFNumberGetValue(window_id_ref, kCFNumberIntType, &window_id); + if (IsWindowFullScreen(desktop_config, window)) { + full_screen_window = window_id; + break; + } + } + + CFRelease(window_array); + return full_screen_window; +} + +bool IsChromeWindow(CGWindowID id) { + int pid = GetWindowOwnerPid(id); + char buffer[PROC_PIDPATHINFO_MAXSIZE]; + int path_length = proc_pidpath(pid, buffer, sizeof(buffer)); + if (path_length <= 0) + return false; + + const char* last_slash = strrchr(buffer, '/'); + std::string name(last_slash ? last_slash + 1 : buffer); + return name.find("Google Chrome") == 0 || name == "Chromium"; +} + +} // namespace + +FullScreenChromeWindowDetector::FullScreenChromeWindowDetector() + : ref_count_(0) {} + +FullScreenChromeWindowDetector::~FullScreenChromeWindowDetector() {} + +CGWindowID FullScreenChromeWindowDetector::FindFullScreenWindow( + CGWindowID original_window) { + if (!IsChromeWindow(original_window) || !IsWindowMinimized(original_window)) + return kCGNullWindowID; + + CGWindowID full_screen_window_id = + FindFullScreenWindowWithSamePidAndTitle(original_window); + + if (full_screen_window_id == kCGNullWindowID) + return kCGNullWindowID; + + for (WindowCapturer::WindowList::iterator it = previous_window_list_.begin(); + it != previous_window_list_.end(); ++it) { + if (static_cast<CGWindowID>(it->id) != full_screen_window_id) + continue; + + int64_t time_interval = + (TickTime::Now() - last_udpate_time_).Milliseconds(); + LOG(LS_WARNING) << "The full-screen window exists in the list, " + << "which was updated " << time_interval << "ms ago."; + return kCGNullWindowID; + } + + return full_screen_window_id; +} + +void FullScreenChromeWindowDetector::UpdateWindowListIfNeeded( + CGWindowID original_window) { + if (IsChromeWindow(original_window) && + (TickTime::Now() - last_udpate_time_).Milliseconds() + > kUpdateIntervalMs) { + previous_window_list_.clear(); + previous_window_list_.swap(current_window_list_); + + // No need to update the window list when the window is minimized. + if (IsWindowMinimized(original_window)) { + previous_window_list_.clear(); + return; + } + + GetWindowList(¤t_window_list_); + last_udpate_time_ = TickTime::Now(); + } +} + +} // namespace webrtc diff --git a/modules/desktop_capture/mac/full_screen_chrome_window_detector.h b/modules/desktop_capture/mac/full_screen_chrome_window_detector.h new file mode 100644 index 00000000..b24fc997 --- /dev/null +++ b/modules/desktop_capture/mac/full_screen_chrome_window_detector.h @@ -0,0 +1,69 @@ +/* + * Copyright (c) 2014 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_DESKTOP_CAPTURE_MAC_FULL_SCREEN_CHROME_WINDOW_DETECTOR_H_ +#define WEBRTC_MODULES_DESKTOP_CAPTURE_MAC_FULL_SCREEN_CHROME_WINDOW_DETECTOR_H_ + +#include <ApplicationServices/ApplicationServices.h> + +#include "webrtc/modules/desktop_capture/window_capturer.h" +#include "webrtc/system_wrappers/interface/atomic32.h" +#include "webrtc/system_wrappers/interface/tick_util.h" + +namespace webrtc { + +// This is a work around for the Chrome tab full-screen behavior: Chrome +// creates a new window in full-screen mode to show a tab full-screen and +// minimizes the old window. To continue capturing in this case, we try to +// find the new full-screen window using these criteria: +// 0. The original shared window is minimized. +// 1. The original shared window's owner application name is "Google Chrome". +// 2. The original window and the new window have the same title and owner +// pid. +// 3. The new window is full-screen. +// 4. The new window didn't exist at least 500 millisecond ago. + +class FullScreenChromeWindowDetector { + public: + FullScreenChromeWindowDetector(); + + void AddRef() { ++ref_count_; } + void Release() { + if (--ref_count_ == 0) + delete this; + } + + // Returns the full-screen window in place of the original window if all the + // criteria are met, or kCGNullWindowID if no such window found. + CGWindowID FindFullScreenWindow(CGWindowID original_window); + + // The caller should call this function periodically, no less than twice per + // second. + void UpdateWindowListIfNeeded(CGWindowID original_window); + + private: + ~FullScreenChromeWindowDetector(); + + Atomic32 ref_count_; + + // We cache the last two results of the window list, so + // |previous_window_list_| is taken at least 500ms before the next Capture() + // call. If we only save the last result, we may get false positive (i.e. + // full-screen window exists in the list) if Capture() is called too soon. + WindowCapturer::WindowList current_window_list_; + WindowCapturer::WindowList previous_window_list_; + TickTime last_udpate_time_; + + DISALLOW_COPY_AND_ASSIGN(FullScreenChromeWindowDetector); +}; + +} // namespace webrtc + +#endif // WEBRTC_MODULES_DESKTOP_CAPTURE_MAC_FULL_SCREEN_CHROME_WINDOW_DETECTOR_H_ diff --git a/modules/desktop_capture/mac/osx_version.cc b/modules/desktop_capture/mac/osx_version.cc deleted file mode 100644 index 7466f203..00000000 --- a/modules/desktop_capture/mac/osx_version.cc +++ /dev/null @@ -1,54 +0,0 @@ -/* - * Copyright (c) 2014 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 <sys/utsname.h> - -#include "webrtc/system_wrappers/interface/logging.h" - -namespace webrtc { - -namespace { - -int GetDarwinVersion() { - struct utsname uname_info; - if (uname(&uname_info) != 0) { - LOG(LS_ERROR) << "uname failed"; - return 0; - } - - if (strcmp(uname_info.sysname, "Darwin") != 0) - return 0; - - char* dot; - int result = strtol(uname_info.release, &dot, 10); - if (*dot != '.') { - LOG(LS_ERROR) << "Failed to parse version"; - return 0; - } - - return result; -} - -} // namespace - -bool IsOSLionOrLater() { - static int darwin_version = GetDarwinVersion(); - - // Verify that the version has been parsed correctly. - if (darwin_version < 6) { - LOG_F(LS_ERROR) << "Invalid Darwin version: " << darwin_version; - abort(); - } - - // Darwin major version 11 corresponds to OSX 10.7. - return darwin_version >= 11; -} - -} // namespace webrtc diff --git a/modules/desktop_capture/mac/window_list_utils.cc b/modules/desktop_capture/mac/window_list_utils.cc new file mode 100644 index 00000000..0c3eaa3a --- /dev/null +++ b/modules/desktop_capture/mac/window_list_utils.cc @@ -0,0 +1,62 @@ +/* + * Copyright (c) 2014 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/desktop_capture/mac/window_list_utils.h" + +#include <ApplicationServices/ApplicationServices.h> + +#include "webrtc/base/macutils.h" + +namespace webrtc { + +bool GetWindowList(WindowCapturer::WindowList* windows) { + // Only get on screen, non-desktop windows. + CFArrayRef window_array = CGWindowListCopyWindowInfo( + kCGWindowListOptionOnScreenOnly | kCGWindowListExcludeDesktopElements, + kCGNullWindowID); + if (!window_array) + return false; + + // Check windows to make sure they have an id, title, and use window layer + // other than 0. + CFIndex count = CFArrayGetCount(window_array); + for (CFIndex i = 0; i < count; ++i) { + CFDictionaryRef window = reinterpret_cast<CFDictionaryRef>( + CFArrayGetValueAtIndex(window_array, i)); + CFStringRef window_title = reinterpret_cast<CFStringRef>( + CFDictionaryGetValue(window, kCGWindowName)); + CFNumberRef window_id = reinterpret_cast<CFNumberRef>( + CFDictionaryGetValue(window, kCGWindowNumber)); + CFNumberRef window_layer = reinterpret_cast<CFNumberRef>( + CFDictionaryGetValue(window, kCGWindowLayer)); + if (window_title && window_id && window_layer) { + // Skip windows with layer=0 (menu, dock). + int layer; + CFNumberGetValue(window_layer, kCFNumberIntType, &layer); + if (layer != 0) + continue; + + int id; + CFNumberGetValue(window_id, kCFNumberIntType, &id); + WindowCapturer::Window window; + window.id = id; + if (!rtc::ToUtf8(window_title, &(window.title)) || + window.title.empty()) { + continue; + } + windows->push_back(window); + } + } + + CFRelease(window_array); + return true; +} + +} // namespace webrtc diff --git a/modules/desktop_capture/mac/osx_version.h b/modules/desktop_capture/mac/window_list_utils.h index 0ba49a4e..7be38506 100644 --- a/modules/desktop_capture/mac/osx_version.h +++ b/modules/desktop_capture/mac/window_list_utils.h @@ -8,9 +8,17 @@ * be found in the AUTHORS file in the root of the source tree. */ +#ifndef WEBRTC_MODULES_DESKTOP_CAPTURE_WINDOW_LIST_UTILS_H_ +#define WEBRTC_MODULES_DESKTOP_CAPTURE_WINDOW_LIST_UTILS_H_ + +#include "webrtc/modules/desktop_capture/window_capturer.h" + namespace webrtc { -// Returns true if the OS version >= OSX 10.7. -bool IsOSLionOrLater(); +// A helper function to get the on-screen windows. +bool GetWindowList(WindowCapturer::WindowList* windows); } // namespace webrtc + +#endif // WEBRTC_MODULES_DESKTOP_CAPTURE_WINDOW_LIST_UTILS_H_ + diff --git a/modules/desktop_capture/mouse_cursor_monitor_mac.mm b/modules/desktop_capture/mouse_cursor_monitor_mac.mm index e8806338..f33720d1 100644 --- a/modules/desktop_capture/mouse_cursor_monitor_mac.mm +++ b/modules/desktop_capture/mouse_cursor_monitor_mac.mm @@ -15,11 +15,12 @@ #include <Cocoa/Cocoa.h> #include <CoreFoundation/CoreFoundation.h> +#include "webrtc/base/macutils.h" #include "webrtc/modules/desktop_capture/desktop_capture_options.h" #include "webrtc/modules/desktop_capture/desktop_frame.h" #include "webrtc/modules/desktop_capture/mac/desktop_configuration.h" #include "webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h" -#include "webrtc/modules/desktop_capture/mac/osx_version.h" +#include "webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.h" #include "webrtc/modules/desktop_capture/mouse_cursor.h" #include "webrtc/system_wrappers/interface/logging.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" @@ -52,6 +53,8 @@ class MouseCursorMonitorMac : public MouseCursorMonitor { Callback* callback_; Mode mode_; scoped_ptr<MouseCursor> last_cursor_; + scoped_refptr<FullScreenChromeWindowDetector> + full_screen_chrome_window_detector_; }; MouseCursorMonitorMac::MouseCursorMonitorMac( @@ -62,9 +65,12 @@ MouseCursorMonitorMac::MouseCursorMonitorMac( window_id_(window_id), screen_id_(screen_id), callback_(NULL), - mode_(SHAPE_AND_POSITION) { + mode_(SHAPE_AND_POSITION), + full_screen_chrome_window_detector_( + options.full_screen_chrome_window_detector()) { assert(window_id == kCGNullWindowID || screen_id == kInvalidScreenId); - if (screen_id != kInvalidScreenId && !IsOSLionOrLater()) { + if (screen_id != kInvalidScreenId && + rtc::GetOSVersionName() < rtc::kMacOSLion) { // Single screen capture is not supported on pre OS X 10.7. screen_id_ = kFullDesktopScreenId; } @@ -115,14 +121,23 @@ void MouseCursorMonitorMac::Capture() { // if the current mouse position is covered by another window and also adjust // |position| to make it relative to the window origin. if (window_id_ != kCGNullWindowID) { - // Get list of windows that may be covering parts of |window_id_|. + CGWindowID on_screen_window = window_id_; + if (full_screen_chrome_window_detector_) { + CGWindowID full_screen_window = + full_screen_chrome_window_detector_->FindFullScreenWindow(window_id_); + + if (full_screen_window != kCGNullWindowID) + on_screen_window = full_screen_window; + } + + // Get list of windows that may be covering parts of |on_screen_window|. // CGWindowListCopyWindowInfo() returns windows in order from front to back, - // so |window_id_| is expected to be the last in the list. + // so |on_screen_window| is expected to be the last in the list. CFArrayRef window_array = CGWindowListCopyWindowInfo(kCGWindowListOptionOnScreenOnly | kCGWindowListOptionOnScreenAboveWindow | kCGWindowListOptionIncludingWindow, - window_id_); + on_screen_window); bool found_window = false; if (window_array) { CFIndex count = CFArrayGetCount(window_array); @@ -158,7 +173,7 @@ void MouseCursorMonitorMac::Capture() { if (!CFNumberGetValue(window_number, kCFNumberIntType, &window_id)) continue; - if (window_id == window_id_) { + if (window_id == on_screen_window) { found_window = true; if (!window_rect.Contains(position)) state = OUTSIDE; diff --git a/modules/desktop_capture/screen_capturer_mac.mm b/modules/desktop_capture/screen_capturer_mac.mm index 2d573390..be05bd99 100644 --- a/modules/desktop_capture/screen_capturer_mac.mm +++ b/modules/desktop_capture/screen_capturer_mac.mm @@ -20,13 +20,13 @@ #include <OpenGL/CGLMacro.h> #include <OpenGL/OpenGL.h> +#include "webrtc/base/macutils.h" #include "webrtc/modules/desktop_capture/desktop_capture_options.h" #include "webrtc/modules/desktop_capture/desktop_frame.h" #include "webrtc/modules/desktop_capture/desktop_geometry.h" #include "webrtc/modules/desktop_capture/desktop_region.h" #include "webrtc/modules/desktop_capture/mac/desktop_configuration.h" #include "webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h" -#include "webrtc/modules/desktop_capture/mac/osx_version.h" #include "webrtc/modules/desktop_capture/mac/scoped_pixel_buffer_object.h" #include "webrtc/modules/desktop_capture/mouse_cursor_shape.h" #include "webrtc/modules/desktop_capture/screen_capture_frame_queue.h" @@ -425,7 +425,7 @@ void ScreenCapturerMac::Capture(const DesktopRegion& region_to_capture) { DesktopFrame* current_frame = queue_.current_frame(); bool flip = false; // GL capturers need flipping. - if (IsOSLionOrLater()) { + if (rtc::GetOSVersionName() >= rtc::kMacOSLion) { // Lion requires us to use their new APIs for doing screen capture. These // APIS currently crash on 10.6.8 if there is no monitor attached. if (!CgBlitPostLion(*current_frame, region)) { @@ -478,7 +478,7 @@ void ScreenCapturerMac::SetMouseShapeObserver( bool ScreenCapturerMac::GetScreenList(ScreenList* screens) { assert(screens->size() == 0); - if (!IsOSLionOrLater()) { + if (rtc::GetOSVersionName() < rtc::kMacOSLion) { // Single monitor cast is not supported on pre OS X 10.7. Screen screen; screen.id = kFullDesktopScreenId; @@ -496,7 +496,7 @@ bool ScreenCapturerMac::GetScreenList(ScreenList* screens) { } bool ScreenCapturerMac::SelectScreen(ScreenId id) { - if (!IsOSLionOrLater()) { + if (rtc::GetOSVersionName() < rtc::kMacOSLion) { // Ignore the screen selection on unsupported OS. assert(!current_display_); return id == kFullDesktopScreenId; @@ -874,7 +874,7 @@ void ScreenCapturerMac::ScreenConfigurationChanged() { // contents. Although the API exists in OS 10.6, it crashes the caller if // the machine has no monitor connected, so we fall back to depcreated APIs // when running on 10.6. - if (IsOSLionOrLater()) { + if (rtc::GetOSVersionName() >= rtc::kMacOSLion) { LOG(LS_INFO) << "Using CgBlitPostLion."; // No need for any OpenGL support on Lion return; @@ -922,10 +922,11 @@ void ScreenCapturerMac::ScreenConfigurationChanged() { LOG(LS_INFO) << "Using GlBlit"; CGLPixelFormatAttribute attributes[] = { - // This function does an early return if IsOSLionOrLater(), this code only - // runs on 10.6 and can be deleted once 10.6 support is dropped. So just - // keep using kCGLPFAFullScreen even though it was deprecated in 10.6 -- - // it's still functional there, and it's not used on newer OS X versions. + // This function does an early return if GetOSVersionName() >= kMacOSLion, + // this code only runs on 10.6 and can be deleted once 10.6 support is + // dropped. So just keep using kCGLPFAFullScreen even though it was + // deprecated in 10.6 -- it's still functional there, and it's not used on + // newer OS X versions. #pragma clang diagnostic push #pragma clang diagnostic ignored "-Wdeprecated-declarations" kCGLPFAFullScreen, diff --git a/modules/desktop_capture/window_capturer_mac.mm b/modules/desktop_capture/window_capturer_mac.mm index d177fc40..f60be5d6 100644 --- a/modules/desktop_capture/window_capturer_mac.mm +++ b/modules/desktop_capture/window_capturer_mac.mm @@ -15,33 +15,21 @@ #include <Cocoa/Cocoa.h> #include <CoreFoundation/CoreFoundation.h> +#include "webrtc/base/macutils.h" +#include "webrtc/modules/desktop_capture/desktop_capture_options.h" #include "webrtc/modules/desktop_capture/desktop_frame.h" +#include "webrtc/modules/desktop_capture/mac/desktop_configuration.h" +#include "webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.h" +#include "webrtc/modules/desktop_capture/mac/window_list_utils.h" #include "webrtc/system_wrappers/interface/logging.h" +#include "webrtc/system_wrappers/interface/scoped_refptr.h" +#include "webrtc/system_wrappers/interface/tick_util.h" namespace webrtc { namespace { -bool CFStringRefToUtf8(const CFStringRef string, std::string* str_utf8) { - assert(string); - assert(str_utf8); - CFIndex length = CFStringGetLength(string); - size_t max_length_utf8 = - CFStringGetMaximumSizeForEncoding(length, kCFStringEncodingUTF8); - str_utf8->resize(max_length_utf8); - CFIndex used_bytes; - int result = CFStringGetBytes( - string, CFRangeMake(0, length), kCFStringEncodingUTF8, 0, false, - reinterpret_cast<UInt8*>(&*str_utf8->begin()), max_length_utf8, - &used_bytes); - if (result != length) { - str_utf8->clear(); - return false; - } - str_utf8->resize(used_bytes); - return true; -} - +// Returns true if the window exists. bool IsWindowValid(CGWindowID id) { CFArrayRef window_id_array = CFArrayCreate(NULL, reinterpret_cast<const void **>(&id), 1, NULL); @@ -56,7 +44,9 @@ bool IsWindowValid(CGWindowID id) { class WindowCapturerMac : public WindowCapturer { public: - WindowCapturerMac(); + explicit WindowCapturerMac( + scoped_refptr<FullScreenChromeWindowDetector> + full_screen_chrome_window_detector); virtual ~WindowCapturerMac(); // WindowCapturer interface. @@ -70,14 +60,22 @@ class WindowCapturerMac : public WindowCapturer { private: Callback* callback_; + + // The window being captured. CGWindowID window_id_; + scoped_refptr<FullScreenChromeWindowDetector> + full_screen_chrome_window_detector_; + DISALLOW_COPY_AND_ASSIGN(WindowCapturerMac); }; -WindowCapturerMac::WindowCapturerMac() +WindowCapturerMac::WindowCapturerMac( + scoped_refptr<FullScreenChromeWindowDetector> + full_screen_chrome_window_detector) : callback_(NULL), - window_id_(0) { + window_id_(0), + full_screen_chrome_window_detector_(full_screen_chrome_window_detector) { } WindowCapturerMac::~WindowCapturerMac() { @@ -114,7 +112,7 @@ bool WindowCapturerMac::GetWindowList(WindowList* windows) { CFNumberGetValue(window_id, kCFNumberIntType, &id); WindowCapturer::Window window; window.id = id; - if (!CFStringRefToUtf8(window_title, &(window.title)) || + if (!rtc::ToUtf8(window_title, &(window.title)) || window.title.empty()) { continue; } @@ -183,9 +181,18 @@ void WindowCapturerMac::Capture(const DesktopRegion& region) { return; } + CGWindowID on_screen_window = window_id_; + if (full_screen_chrome_window_detector_) { + CGWindowID full_screen_window = + full_screen_chrome_window_detector_->FindFullScreenWindow(window_id_); + + if (full_screen_window != kCGNullWindowID) + on_screen_window = full_screen_window; + } + CGImageRef window_image = CGWindowListCreateImage( CGRectNull, kCGWindowListOptionIncludingWindow, - window_id_, kCGWindowImageBoundsIgnoreFraming); + on_screen_window, kCGWindowImageBoundsIgnoreFraming); if (!window_image) { callback_->OnCaptureCompleted(NULL); @@ -218,13 +225,16 @@ void WindowCapturerMac::Capture(const DesktopRegion& region) { CFRelease(window_image); callback_->OnCaptureCompleted(frame); + + if (full_screen_chrome_window_detector_) + full_screen_chrome_window_detector_->UpdateWindowListIfNeeded(window_id_); } } // namespace // static WindowCapturer* WindowCapturer::Create(const DesktopCaptureOptions& options) { - return new WindowCapturerMac(); + return new WindowCapturerMac(options.full_screen_chrome_window_detector()); } } // namespace webrtc diff --git a/modules/interface/module_common_types.h b/modules/interface/module_common_types.h index 2c947071..e37313c9 100644 --- a/modules/interface/module_common_types.h +++ b/modules/interface/module_common_types.h @@ -75,14 +75,22 @@ struct RTPVideoHeaderVP8 { bool beginningOfPartition; // True if this packet is the first // in a VP8 partition. Otherwise false }; + +struct RTPVideoHeaderH264 { + uint8_t nalu_header; + bool single_nalu; +}; + union RTPVideoTypeHeader { RTPVideoHeaderVP8 VP8; + RTPVideoHeaderH264 H264; }; enum RtpVideoCodecTypes { kRtpVideoNone, kRtpVideoGeneric, - kRtpVideoVp8 + kRtpVideoVp8, + kRtpVideoH264 }; struct RTPVideoHeader { uint16_t width; // size diff --git a/modules/pacing/include/mock/mock_paced_sender.h b/modules/pacing/include/mock/mock_paced_sender.h index 3841ef37..6600a929 100644 --- a/modules/pacing/include/mock/mock_paced_sender.h +++ b/modules/pacing/include/mock/mock_paced_sender.h @@ -16,12 +16,13 @@ #include <vector> #include "webrtc/modules/pacing/include/paced_sender.h" +#include "webrtc/system_wrappers/interface/clock.h" namespace webrtc { class MockPacedSender : public PacedSender { public: - MockPacedSender() : PacedSender(NULL, 0, 0) {} + MockPacedSender() : PacedSender(Clock::GetRealTimeClock(), NULL, 0, 0) {} MOCK_METHOD6(SendPacket, bool(Priority priority, uint32_t ssrc, uint16_t sequence_number, diff --git a/modules/pacing/include/paced_sender.h b/modules/pacing/include/paced_sender.h index 95f1a86e..55497db3 100644 --- a/modules/pacing/include/paced_sender.h +++ b/modules/pacing/include/paced_sender.h @@ -16,11 +16,14 @@ #include "webrtc/modules/interface/module.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" +#include "webrtc/system_wrappers/interface/thread_annotations.h" #include "webrtc/system_wrappers/interface/tick_util.h" #include "webrtc/typedefs.h" namespace webrtc { +class Clock; class CriticalSectionWrapper; + namespace paced_sender { class IntervalBudget; struct Packet; @@ -48,6 +51,7 @@ class PacedSender : public Module { int64_t capture_time_ms, bool retransmission) = 0; // Called when it's a good time to send a padding data. + // Returns the number of bytes sent. virtual int TimeToSendPadding(int bytes) = 0; protected: @@ -55,8 +59,17 @@ class PacedSender : public Module { }; static const int kDefaultMaxQueueLengthMs = 2000; - - PacedSender(Callback* callback, int max_bitrate_kbps, int min_bitrate_kbps); + // Pace in kbits/s until we receive first estimate. + static const int kDefaultInitialPaceKbps = 2000; + // Pacing-rate relative to our target send rate. + // Multiplicative factor that is applied to the target bitrate to calculate + // the number of bytes that can be transmitted per interval. + // Increasing this factor will result in lower delays in cases of bitrate + // overshoots from the encoder. + static const float kDefaultPaceMultiplier; + + PacedSender(Clock* clock, Callback* callback, int max_bitrate_kbps, + int min_bitrate_kbps); virtual ~PacedSender(); @@ -101,40 +114,50 @@ class PacedSender : public Module { private: // Return true if next packet in line should be transmitted. // Return packet list that contains the next packet. - bool ShouldSendNextPacket(paced_sender::PacketList** packet_list); + bool ShouldSendNextPacket(paced_sender::PacketList** packet_list) + EXCLUSIVE_LOCKS_REQUIRED(critsect_); // Local helper function to GetNextPacket. - paced_sender::Packet GetNextPacketFromList(paced_sender::PacketList* packets); + paced_sender::Packet GetNextPacketFromList(paced_sender::PacketList* packets) + EXCLUSIVE_LOCKS_REQUIRED(critsect_); - bool SendPacketFromList(paced_sender::PacketList* packet_list); + bool SendPacketFromList(paced_sender::PacketList* packet_list) + EXCLUSIVE_LOCKS_REQUIRED(critsect_); // Updates the number of bytes that can be sent for the next time interval. - void UpdateBytesPerInterval(uint32_t delta_time_in_ms); + void UpdateBytesPerInterval(uint32_t delta_time_in_ms) + EXCLUSIVE_LOCKS_REQUIRED(critsect_); // Updates the buffers with the number of bytes that we sent. - void UpdateMediaBytesSent(int num_bytes); + void UpdateMediaBytesSent(int num_bytes) EXCLUSIVE_LOCKS_REQUIRED(critsect_); + + Clock* const clock_; + Callback* const callback_; - Callback* callback_; - bool enabled_; - bool paused_; - int max_queue_length_ms_; scoped_ptr<CriticalSectionWrapper> critsect_; + bool enabled_ GUARDED_BY(critsect_); + bool paused_ GUARDED_BY(critsect_); + int max_queue_length_ms_ GUARDED_BY(critsect_); // This is the media budget, keeping track of how many bits of media // we can pace out during the current interval. - scoped_ptr<paced_sender::IntervalBudget> media_budget_; + scoped_ptr<paced_sender::IntervalBudget> media_budget_ GUARDED_BY(critsect_); // This is the padding budget, keeping track of how many bits of padding we're // allowed to send out during the current interval. This budget will be // utilized when there's no media to send. - scoped_ptr<paced_sender::IntervalBudget> padding_budget_; - - TickTime time_last_update_; - TickTime time_last_send_; - int64_t capture_time_ms_last_queued_; - int64_t capture_time_ms_last_sent_; - - scoped_ptr<paced_sender::PacketList> high_priority_packets_; - scoped_ptr<paced_sender::PacketList> normal_priority_packets_; - scoped_ptr<paced_sender::PacketList> low_priority_packets_; + scoped_ptr<paced_sender::IntervalBudget> padding_budget_ + GUARDED_BY(critsect_); + + TickTime time_last_update_ GUARDED_BY(critsect_); + TickTime time_last_send_ GUARDED_BY(critsect_); + int64_t capture_time_ms_last_queued_ GUARDED_BY(critsect_); + int64_t capture_time_ms_last_sent_ GUARDED_BY(critsect_); + + scoped_ptr<paced_sender::PacketList> high_priority_packets_ + GUARDED_BY(critsect_); + scoped_ptr<paced_sender::PacketList> normal_priority_packets_ + GUARDED_BY(critsect_); + scoped_ptr<paced_sender::PacketList> low_priority_packets_ + GUARDED_BY(critsect_); }; } // namespace webrtc #endif // WEBRTC_MODULES_PACED_SENDER_H_ diff --git a/modules/pacing/paced_sender.cc b/modules/pacing/paced_sender.cc index e9f9bddc..323cafec 100644 --- a/modules/pacing/paced_sender.cc +++ b/modules/pacing/paced_sender.cc @@ -12,7 +12,11 @@ #include <assert.h> +#include <map> +#include <set> + #include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/system_wrappers/interface/clock.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/trace_event.h" @@ -35,21 +39,24 @@ namespace webrtc { namespace paced_sender { struct Packet { - Packet(uint32_t ssrc, uint16_t seq_number, int64_t capture_time_ms, - int64_t enqueue_time_ms, int length_in_bytes, bool retransmission) - : ssrc_(ssrc), - sequence_number_(seq_number), - capture_time_ms_(capture_time_ms), - enqueue_time_ms_(enqueue_time_ms), - bytes_(length_in_bytes), - retransmission_(retransmission) { - } - uint32_t ssrc_; - uint16_t sequence_number_; - int64_t capture_time_ms_; - int64_t enqueue_time_ms_; - int bytes_; - bool retransmission_; + Packet(uint32_t ssrc, + uint16_t seq_number, + int64_t capture_time_ms, + int64_t enqueue_time_ms, + int length_in_bytes, + bool retransmission) + : ssrc(ssrc), + sequence_number(seq_number), + capture_time_ms(capture_time_ms), + enqueue_time_ms(enqueue_time_ms), + bytes(length_in_bytes), + retransmission(retransmission) {} + uint32_t ssrc; + uint16_t sequence_number; + int64_t capture_time_ms; + int64_t enqueue_time_ms; + int bytes; + bool retransmission; }; // STL list style class which prevents duplicates in the list. @@ -67,23 +74,24 @@ class PacketList { void pop_front() { Packet& packet = packet_list_.front(); - uint16_t sequence_number = packet.sequence_number_; + uint16_t sequence_number = packet.sequence_number; + uint32_t ssrc = packet.ssrc; packet_list_.pop_front(); - sequence_number_set_.erase(sequence_number); + sequence_number_set_[ssrc].erase(sequence_number); } void push_back(const Packet& packet) { - if (sequence_number_set_.find(packet.sequence_number_) == - sequence_number_set_.end()) { + if (sequence_number_set_[packet.ssrc].find(packet.sequence_number) == + sequence_number_set_[packet.ssrc].end()) { // Don't insert duplicates. packet_list_.push_back(packet); - sequence_number_set_.insert(packet.sequence_number_); + sequence_number_set_[packet.ssrc].insert(packet.sequence_number); } } private: std::list<Packet> packet_list_; - std::set<uint16_t> sequence_number_set_; + std::map<uint32_t, std::set<uint16_t> > sequence_number_set_; }; class IntervalBudget { @@ -120,14 +128,18 @@ class IntervalBudget { }; } // namespace paced_sender -PacedSender::PacedSender(Callback* callback, +const float PacedSender::kDefaultPaceMultiplier = 2.5f; + +PacedSender::PacedSender(Clock* clock, + Callback* callback, int max_bitrate_kbps, int min_bitrate_kbps) - : callback_(callback), + : clock_(clock), + callback_(callback), + critsect_(CriticalSectionWrapper::CreateCriticalSection()), enabled_(true), paused_(false), max_queue_length_ms_(kDefaultMaxQueueLengthMs), - critsect_(CriticalSectionWrapper::CreateCriticalSection()), media_budget_(new paced_sender::IntervalBudget(max_bitrate_kbps)), padding_budget_(new paced_sender::IntervalBudget(min_bitrate_kbps)), time_last_update_(TickTime::Now()), @@ -178,7 +190,7 @@ bool PacedSender::SendPacket(Priority priority, uint32_t ssrc, return true; // We can send now. } if (capture_time_ms < 0) { - capture_time_ms = TickTime::MillisecondTimestamp(); + capture_time_ms = clock_->TimeInMilliseconds(); } if (priority != kHighPriority && capture_time_ms > capture_time_ms_last_queued_) { @@ -201,7 +213,7 @@ bool PacedSender::SendPacket(Priority priority, uint32_t ssrc, packet_list->push_back(paced_sender::Packet(ssrc, sequence_number, capture_time_ms, - TickTime::MillisecondTimestamp(), + clock_->TimeInMilliseconds(), bytes, retransmission)); return false; @@ -214,22 +226,22 @@ void PacedSender::set_max_queue_length_ms(int max_queue_length_ms) { int PacedSender::QueueInMs() const { CriticalSectionScoped cs(critsect_.get()); - int64_t now_ms = TickTime::MillisecondTimestamp(); + int64_t now_ms = clock_->TimeInMilliseconds(); int64_t oldest_packet_enqueue_time = now_ms; if (!high_priority_packets_->empty()) { - oldest_packet_enqueue_time = std::min( - oldest_packet_enqueue_time, - high_priority_packets_->front().enqueue_time_ms_); + oldest_packet_enqueue_time = + std::min(oldest_packet_enqueue_time, + high_priority_packets_->front().enqueue_time_ms); } if (!normal_priority_packets_->empty()) { - oldest_packet_enqueue_time = std::min( - oldest_packet_enqueue_time, - normal_priority_packets_->front().enqueue_time_ms_); + oldest_packet_enqueue_time = + std::min(oldest_packet_enqueue_time, + normal_priority_packets_->front().enqueue_time_ms); } if (!low_priority_packets_->empty()) { - oldest_packet_enqueue_time = std::min( - oldest_packet_enqueue_time, - low_priority_packets_->front().enqueue_time_ms_); + oldest_packet_enqueue_time = + std::min(oldest_packet_enqueue_time, + low_priority_packets_->front().enqueue_time_ms); } return now_ms - oldest_packet_enqueue_time; } @@ -286,10 +298,10 @@ bool PacedSender::SendPacketFromList(paced_sender::PacketList* packet_list) paced_sender::Packet packet = GetNextPacketFromList(packet_list); critsect_->Leave(); - const bool success = callback_->TimeToSendPacket(packet.ssrc_, - packet.sequence_number_, - packet.capture_time_ms_, - packet.retransmission_); + const bool success = callback_->TimeToSendPacket(packet.ssrc, + packet.sequence_number, + packet.capture_time_ms, + packet.retransmission); critsect_->Enter(); // If packet cannot be sent then keep it in packet list and exit early. // There's no need to send more packets. @@ -297,15 +309,15 @@ bool PacedSender::SendPacketFromList(paced_sender::PacketList* packet_list) return false; } packet_list->pop_front(); - const bool last_packet = packet_list->empty() || - packet_list->front().capture_time_ms_ > packet.capture_time_ms_; + const bool last_packet = + packet_list->empty() || + packet_list->front().capture_time_ms > packet.capture_time_ms; if (packet_list != high_priority_packets_.get()) { - if (packet.capture_time_ms_ > capture_time_ms_last_sent_) { - capture_time_ms_last_sent_ = packet.capture_time_ms_; - } else if (packet.capture_time_ms_ == capture_time_ms_last_sent_ && + if (packet.capture_time_ms > capture_time_ms_last_sent_) { + capture_time_ms_last_sent_ = packet.capture_time_ms; + } else if (packet.capture_time_ms == capture_time_ms_last_sent_ && last_packet) { - TRACE_EVENT_ASYNC_END0("webrtc_rtp", "PacedSend", - packet.capture_time_ms_); + TRACE_EVENT_ASYNC_END0("webrtc_rtp", "PacedSend", packet.capture_time_ms); } } return true; @@ -339,12 +351,13 @@ bool PacedSender::ShouldSendNextPacket(paced_sender::PacketList** packet_list) { int64_t high_priority_capture_time = -1; if (!high_priority_packets_->empty()) { high_priority_capture_time = - high_priority_packets_->front().capture_time_ms_; + high_priority_packets_->front().capture_time_ms; *packet_list = high_priority_packets_.get(); } if (!normal_priority_packets_->empty() && - (high_priority_capture_time == -1 || high_priority_capture_time > - normal_priority_packets_->front().capture_time_ms_)) { + (high_priority_capture_time == -1 || + high_priority_capture_time > + normal_priority_packets_->front().capture_time_ms)) { *packet_list = normal_priority_packets_.get(); } if (*packet_list) @@ -370,7 +383,7 @@ bool PacedSender::ShouldSendNextPacket(paced_sender::PacketList** packet_list) { paced_sender::Packet PacedSender::GetNextPacketFromList( paced_sender::PacketList* packets) { paced_sender::Packet packet = packets->front(); - UpdateMediaBytesSent(packet.bytes_); + UpdateMediaBytesSent(packet.bytes); return packet; } diff --git a/modules/pacing/paced_sender_unittest.cc b/modules/pacing/paced_sender_unittest.cc index 9763c800..55188558 100644 --- a/modules/pacing/paced_sender_unittest.cc +++ b/modules/pacing/paced_sender_unittest.cc @@ -12,6 +12,7 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/modules/pacing/include/paced_sender.h" +#include "webrtc/system_wrappers/interface/clock.h" using testing::_; using testing::Return; @@ -55,12 +56,13 @@ class PacedSenderPadding : public PacedSender::Callback { class PacedSenderTest : public ::testing::Test { protected: - PacedSenderTest() { + PacedSenderTest() : clock_(123456) { srand(0); TickTime::UseFakeClock(123456); // Need to initialize PacedSender after we initialize clock. send_bucket_.reset( - new PacedSender(&callback_, kPaceMultiplier * kTargetBitrate, 0)); + new PacedSender( + &clock_, &callback_, kPaceMultiplier * kTargetBitrate, 0)); } void SendAndExpectPacket(PacedSender::Priority priority, @@ -75,6 +77,7 @@ class PacedSenderTest : public ::testing::Test { .WillRepeatedly(Return(true)); } + SimulatedClock clock_; MockPacedSenderCallback callback_; scoped_ptr<PacedSender> send_bucket_; }; @@ -84,19 +87,21 @@ TEST_F(PacedSenderTest, QueuePacket) { uint16_t sequence_number = 1234; // Due to the multiplicative factor we can send 3 packets not 2 packets. SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++, - TickTime::MillisecondTimestamp(), 250, false); + clock_.TimeInMilliseconds(), 250, false); SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++, - TickTime::MillisecondTimestamp(), 250, false); + clock_.TimeInMilliseconds(), 250, false); SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++, - TickTime::MillisecondTimestamp(), 250, false); - int64_t queued_packet_timestamp = TickTime::MillisecondTimestamp(); + clock_.TimeInMilliseconds(), 250, false); + int64_t queued_packet_timestamp = clock_.TimeInMilliseconds(); EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority, ssrc, sequence_number, queued_packet_timestamp, 250, false)); send_bucket_->Process(); EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess()); EXPECT_CALL(callback_, TimeToSendPadding(_)).Times(0); + clock_.AdvanceTimeMilliseconds(4); TickTime::AdvanceFakeClock(4); EXPECT_EQ(1, send_bucket_->TimeUntilNextProcess()); + clock_.AdvanceTimeMilliseconds(1); TickTime::AdvanceFakeClock(1); EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess()); EXPECT_CALL(callback_, TimeToSendPacket( @@ -106,11 +111,11 @@ TEST_F(PacedSenderTest, QueuePacket) { send_bucket_->Process(); sequence_number++; SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++, - TickTime::MillisecondTimestamp(), 250, false); + clock_.TimeInMilliseconds(), 250, false); SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++, - TickTime::MillisecondTimestamp(), 250, false); + clock_.TimeInMilliseconds(), 250, false); EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority, ssrc, - sequence_number++, TickTime::MillisecondTimestamp(), 250, false)); + sequence_number++, clock_.TimeInMilliseconds(), 250, false)); send_bucket_->Process(); } @@ -121,16 +126,17 @@ TEST_F(PacedSenderTest, PaceQueuedPackets) { // Due to the multiplicative factor we can send 3 packets not 2 packets. for (int i = 0; i < 3; ++i) { SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++, - TickTime::MillisecondTimestamp(), 250, false); + clock_.TimeInMilliseconds(), 250, false); } for (int j = 0; j < 30; ++j) { EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority, ssrc, - sequence_number++, TickTime::MillisecondTimestamp(), 250, false)); + sequence_number++, clock_.TimeInMilliseconds(), 250, false)); } send_bucket_->Process(); EXPECT_CALL(callback_, TimeToSendPadding(_)).Times(0); for (int k = 0; k < 10; ++k) { EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess()); + clock_.AdvanceTimeMilliseconds(5); TickTime::AdvanceFakeClock(5); EXPECT_CALL(callback_, TimeToSendPacket(ssrc, _, _, false)) @@ -140,17 +146,18 @@ TEST_F(PacedSenderTest, PaceQueuedPackets) { EXPECT_EQ(0, send_bucket_->Process()); } EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess()); + clock_.AdvanceTimeMilliseconds(5); TickTime::AdvanceFakeClock(5); EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess()); EXPECT_EQ(0, send_bucket_->Process()); SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++, - TickTime::MillisecondTimestamp(), 250, false); + clock_.TimeInMilliseconds(), 250, false); SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++, - TickTime::MillisecondTimestamp(), 250, false); + clock_.TimeInMilliseconds(), 250, false); SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++, - TickTime::MillisecondTimestamp(), 250, false); + clock_.TimeInMilliseconds(), 250, false); EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority, ssrc, - sequence_number, TickTime::MillisecondTimestamp(), 250, false)); + sequence_number, clock_.TimeInMilliseconds(), 250, false)); send_bucket_->Process(); } @@ -162,21 +169,22 @@ TEST_F(PacedSenderTest, PaceQueuedPacketsWithDuplicates) { // Due to the multiplicative factor we can send 3 packets not 2 packets. for (int i = 0; i < 3; ++i) { SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++, - TickTime::MillisecondTimestamp(), 250, false); + clock_.TimeInMilliseconds(), 250, false); } queued_sequence_number = sequence_number; for (int j = 0; j < 30; ++j) { // Send in duplicate packets. EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority, ssrc, - sequence_number, TickTime::MillisecondTimestamp(), 250, false)); + sequence_number, clock_.TimeInMilliseconds(), 250, false)); EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority, ssrc, - sequence_number++, TickTime::MillisecondTimestamp(), 250, false)); + sequence_number++, clock_.TimeInMilliseconds(), 250, false)); } EXPECT_CALL(callback_, TimeToSendPadding(_)).Times(0); send_bucket_->Process(); for (int k = 0; k < 10; ++k) { EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess()); + clock_.AdvanceTimeMilliseconds(5); TickTime::AdvanceFakeClock(5); for (int i = 0; i < 3; ++i) { @@ -190,17 +198,42 @@ TEST_F(PacedSenderTest, PaceQueuedPacketsWithDuplicates) { EXPECT_EQ(0, send_bucket_->Process()); } EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess()); + clock_.AdvanceTimeMilliseconds(5); TickTime::AdvanceFakeClock(5); EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess()); EXPECT_EQ(0, send_bucket_->Process()); SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++, - TickTime::MillisecondTimestamp(), 250, false); + clock_.TimeInMilliseconds(), 250, false); SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++, - TickTime::MillisecondTimestamp(), 250, false); + clock_.TimeInMilliseconds(), 250, false); SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++, - TickTime::MillisecondTimestamp(), 250, false); + clock_.TimeInMilliseconds(), 250, false); EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority, ssrc, - sequence_number++, TickTime::MillisecondTimestamp(), 250, false)); + sequence_number++, clock_.TimeInMilliseconds(), 250, false)); + send_bucket_->Process(); +} + +TEST_F(PacedSenderTest, CanQueuePacketsWithSameSequenceNumberOnDifferentSsrcs) { + uint32_t ssrc = 12345; + uint16_t sequence_number = 1234; + + SendAndExpectPacket(PacedSender::kNormalPriority, + ssrc, + sequence_number, + clock_.TimeInMilliseconds(), + 250, + false); + + // Expect packet on second ssrc to be queued and sent as well. + SendAndExpectPacket(PacedSender::kNormalPriority, + ssrc + 1, + sequence_number, + clock_.TimeInMilliseconds(), + 250, + false); + + clock_.AdvanceTimeMilliseconds(1000); + TickTime::AdvanceFakeClock(1000); send_bucket_->Process(); } @@ -211,14 +244,15 @@ TEST_F(PacedSenderTest, Padding) { send_bucket_->UpdateBitrate(kPaceMultiplier * kTargetBitrate, kTargetBitrate); // Due to the multiplicative factor we can send 3 packets not 2 packets. SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++, - TickTime::MillisecondTimestamp(), 250, false); + clock_.TimeInMilliseconds(), 250, false); SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++, - TickTime::MillisecondTimestamp(), 250, false); + clock_.TimeInMilliseconds(), 250, false); SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++, - TickTime::MillisecondTimestamp(), 250, false); + clock_.TimeInMilliseconds(), 250, false); // No padding is expected since we have sent too much already. EXPECT_CALL(callback_, TimeToSendPadding(_)).Times(0); EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess()); + clock_.AdvanceTimeMilliseconds(5); TickTime::AdvanceFakeClock(5); EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess()); EXPECT_EQ(0, send_bucket_->Process()); @@ -227,6 +261,7 @@ TEST_F(PacedSenderTest, Padding) { EXPECT_CALL(callback_, TimeToSendPadding(250)).Times(1). WillOnce(Return(250)); EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess()); + clock_.AdvanceTimeMilliseconds(5); TickTime::AdvanceFakeClock(5); EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess()); EXPECT_EQ(0, send_bucket_->Process()); @@ -238,11 +273,13 @@ TEST_F(PacedSenderTest, NoPaddingWhenDisabled) { // No padding is expected since the pacer is disabled. EXPECT_CALL(callback_, TimeToSendPadding(_)).Times(0); EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess()); + clock_.AdvanceTimeMilliseconds(5); TickTime::AdvanceFakeClock(5); EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess()); EXPECT_EQ(0, send_bucket_->Process()); EXPECT_CALL(callback_, TimeToSendPadding(_)).Times(0); EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess()); + clock_.AdvanceTimeMilliseconds(5); TickTime::AdvanceFakeClock(5); EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess()); EXPECT_EQ(0, send_bucket_->Process()); @@ -255,10 +292,11 @@ TEST_F(PacedSenderTest, VerifyPaddingUpToBitrate) { const int kTimeStep = 5; const int64_t kBitrateWindow = 100; send_bucket_->UpdateBitrate(kPaceMultiplier * kTargetBitrate, kTargetBitrate); - int64_t start_time = TickTime::MillisecondTimestamp(); - while (TickTime::MillisecondTimestamp() - start_time < kBitrateWindow) { + int64_t start_time = clock_.TimeInMilliseconds(); + while (clock_.TimeInMilliseconds() - start_time < kBitrateWindow) { SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++, capture_time_ms, 250, false); + clock_.AdvanceTimeMilliseconds(kTimeStep); TickTime::AdvanceFakeClock(kTimeStep); EXPECT_CALL(callback_, TimeToSendPadding(250)).Times(1). WillOnce(Return(250)); @@ -274,16 +312,17 @@ TEST_F(PacedSenderTest, VerifyAverageBitrateVaryingMediaPayload) { const int64_t kBitrateWindow = 10000; PacedSenderPadding callback; send_bucket_.reset( - new PacedSender(&callback, kPaceMultiplier * kTargetBitrate, 0)); + new PacedSender(&clock_, &callback, kPaceMultiplier * kTargetBitrate, 0)); send_bucket_->UpdateBitrate(kPaceMultiplier * kTargetBitrate, kTargetBitrate); - int64_t start_time = TickTime::MillisecondTimestamp(); + int64_t start_time = clock_.TimeInMilliseconds(); int media_bytes = 0; - while (TickTime::MillisecondTimestamp() - start_time < kBitrateWindow) { + while (clock_.TimeInMilliseconds() - start_time < kBitrateWindow) { int media_payload = rand() % 100 + 200; // [200, 300] bytes. EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority, ssrc, sequence_number++, capture_time_ms, media_payload, false)); media_bytes += media_payload; + clock_.AdvanceTimeMilliseconds(kTimeStep); TickTime::AdvanceFakeClock(kTimeStep); send_bucket_->Process(); } @@ -325,6 +364,7 @@ TEST_F(PacedSenderTest, Priority) { .WillRepeatedly(Return(true)); EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess()); + clock_.AdvanceTimeMilliseconds(5); TickTime::AdvanceFakeClock(5); EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess()); EXPECT_EQ(0, send_bucket_->Process()); @@ -335,6 +375,7 @@ TEST_F(PacedSenderTest, Priority) { .WillRepeatedly(Return(true)); EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess()); + clock_.AdvanceTimeMilliseconds(5); TickTime::AdvanceFakeClock(5); EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess()); EXPECT_EQ(0, send_bucket_->Process()); @@ -344,7 +385,7 @@ TEST_F(PacedSenderTest, Pause) { uint32_t ssrc_low_priority = 12345; uint32_t ssrc = 12346; uint16_t sequence_number = 1234; - int64_t capture_time_ms = TickTime::MillisecondTimestamp(); + int64_t capture_time_ms = clock_.TimeInMilliseconds(); EXPECT_EQ(0, send_bucket_->QueueInMs()); @@ -366,15 +407,16 @@ TEST_F(PacedSenderTest, Pause) { EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kHighPriority, ssrc, sequence_number++, capture_time_ms, 250, false)); + clock_.AdvanceTimeMilliseconds(10000); TickTime::AdvanceFakeClock(10000); - int64_t second_capture_time_ms = TickTime::MillisecondTimestamp(); + int64_t second_capture_time_ms = clock_.TimeInMilliseconds(); // Expect everything to be queued. EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kLowPriority, ssrc_low_priority, sequence_number++, second_capture_time_ms, 250, false)); - EXPECT_EQ(TickTime::MillisecondTimestamp() - capture_time_ms, + EXPECT_EQ(clock_.TimeInMilliseconds() - capture_time_ms, send_bucket_->QueueInMs()); // Expect no packet to come out while paused. @@ -382,6 +424,7 @@ TEST_F(PacedSenderTest, Pause) { EXPECT_CALL(callback_, TimeToSendPacket(_, _, _, _)).Times(0); for (int i = 0; i < 10; ++i) { + clock_.AdvanceTimeMilliseconds(5); TickTime::AdvanceFakeClock(5); EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess()); EXPECT_EQ(0, send_bucket_->Process()); @@ -394,6 +437,7 @@ TEST_F(PacedSenderTest, Pause) { send_bucket_->Resume(); EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess()); + clock_.AdvanceTimeMilliseconds(5); TickTime::AdvanceFakeClock(5); EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess()); EXPECT_EQ(0, send_bucket_->Process()); @@ -403,6 +447,7 @@ TEST_F(PacedSenderTest, Pause) { .Times(1) .WillRepeatedly(Return(true)); EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess()); + clock_.AdvanceTimeMilliseconds(5); TickTime::AdvanceFakeClock(5); EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess()); EXPECT_EQ(0, send_bucket_->Process()); @@ -412,7 +457,7 @@ TEST_F(PacedSenderTest, Pause) { TEST_F(PacedSenderTest, ResendPacket) { uint32_t ssrc = 12346; uint16_t sequence_number = 1234; - int64_t capture_time_ms = TickTime::MillisecondTimestamp(); + int64_t capture_time_ms = clock_.TimeInMilliseconds(); EXPECT_EQ(0, send_bucket_->QueueInMs()); EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority, @@ -421,6 +466,7 @@ TEST_F(PacedSenderTest, ResendPacket) { capture_time_ms, 250, false)); + clock_.AdvanceTimeMilliseconds(1); TickTime::AdvanceFakeClock(1); EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority, ssrc, @@ -428,19 +474,21 @@ TEST_F(PacedSenderTest, ResendPacket) { capture_time_ms + 1, 250, false)); + clock_.AdvanceTimeMilliseconds(9999); TickTime::AdvanceFakeClock(9999); - EXPECT_EQ(TickTime::MillisecondTimestamp() - capture_time_ms, + EXPECT_EQ(clock_.TimeInMilliseconds() - capture_time_ms, send_bucket_->QueueInMs()); // Fails to send first packet so only one call. EXPECT_CALL(callback_, TimeToSendPacket( ssrc, sequence_number, capture_time_ms, false)) .Times(1) .WillOnce(Return(false)); + clock_.AdvanceTimeMilliseconds(10000); TickTime::AdvanceFakeClock(10000); send_bucket_->Process(); // Queue remains unchanged. - EXPECT_EQ(TickTime::MillisecondTimestamp() - capture_time_ms, + EXPECT_EQ(clock_.TimeInMilliseconds() - capture_time_ms, send_bucket_->QueueInMs()); // Fails to send second packet. @@ -452,11 +500,12 @@ TEST_F(PacedSenderTest, ResendPacket) { ssrc, sequence_number + 1, capture_time_ms + 1, false)) .Times(1) .WillOnce(Return(false)); + clock_.AdvanceTimeMilliseconds(10000); TickTime::AdvanceFakeClock(10000); send_bucket_->Process(); // Queue is reduced by 1 packet. - EXPECT_EQ(TickTime::MillisecondTimestamp() - capture_time_ms - 1, + EXPECT_EQ(clock_.TimeInMilliseconds() - capture_time_ms - 1, send_bucket_->QueueInMs()); // Send second packet and queue becomes empty. @@ -464,6 +513,7 @@ TEST_F(PacedSenderTest, ResendPacket) { ssrc, sequence_number + 1, capture_time_ms + 1, false)) .Times(1) .WillOnce(Return(true)); + clock_.AdvanceTimeMilliseconds(10000); TickTime::AdvanceFakeClock(10000); send_bucket_->Process(); EXPECT_EQ(0, send_bucket_->QueueInMs()); @@ -479,21 +529,23 @@ TEST_F(PacedSenderTest, MaxQueueLength) { SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++, - TickTime::MillisecondTimestamp(), + clock_.TimeInMilliseconds(), 1200, false); } + clock_.AdvanceTimeMilliseconds(2001); TickTime::AdvanceFakeClock(2001); SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++, - TickTime::MillisecondTimestamp(), + clock_.TimeInMilliseconds(), 1200, false); EXPECT_EQ(2001, send_bucket_->QueueInMs()); send_bucket_->Process(); EXPECT_EQ(0, send_bucket_->QueueInMs()); + clock_.AdvanceTimeMilliseconds(31); TickTime::AdvanceFakeClock(31); send_bucket_->Process(); } @@ -507,10 +559,11 @@ TEST_F(PacedSenderTest, QueueTimeGrowsOverTime) { SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number, - TickTime::MillisecondTimestamp(), + clock_.TimeInMilliseconds(), 1200, false); + clock_.AdvanceTimeMilliseconds(500); TickTime::AdvanceFakeClock(500); EXPECT_EQ(500, send_bucket_->QueueInMs()); send_bucket_->Process(); diff --git a/modules/remote_bitrate_estimator/bwe_simulations.cc b/modules/remote_bitrate_estimator/bwe_simulations.cc index 6b208e49..47390f78 100644 --- a/modules/remote_bitrate_estimator/bwe_simulations.cc +++ b/modules/remote_bitrate_estimator/bwe_simulations.cc @@ -96,6 +96,36 @@ TEST_P(BweSimulation, Choke1000kbps500kbps1000kbps) { RunFor(60 * 1000); } +TEST_P(BweSimulation, PacerChoke1000kbps500kbps1000kbps) { + VerboseLogging(true); + AdaptiveVideoSender source(0, NULL, 30, 300, 0, 0); + PacedVideoSender sender(this, 300, &source); + ChokeFilter filter(this); + RateCounterFilter counter(this, "receiver_input"); + filter.SetCapacity(1000); + filter.SetMaxDelay(500); + RunFor(60 * 1000); + filter.SetCapacity(500); + RunFor(60 * 1000); + filter.SetCapacity(1000); + RunFor(60 * 1000); +} + +TEST_P(BweSimulation, PacerChoke200kbps30kbps200kbps) { + VerboseLogging(true); + AdaptiveVideoSender source(0, NULL, 30, 300, 0, 0); + PacedVideoSender sender(this, 300, &source); + ChokeFilter filter(this); + RateCounterFilter counter(this, "receiver_input"); + filter.SetCapacity(200); + filter.SetMaxDelay(500); + RunFor(60 * 1000); + filter.SetCapacity(30); + RunFor(60 * 1000); + filter.SetCapacity(200); + RunFor(60 * 1000); +} + TEST_P(BweSimulation, Choke200kbps30kbps200kbps) { VerboseLogging(true); AdaptiveVideoSender sender(0, this, 30, 300, 0, 0); diff --git a/modules/remote_bitrate_estimator/test/bwe_test.cc b/modules/remote_bitrate_estimator/test/bwe_test.cc index 90ce6a3e..d4919e73 100644 --- a/modules/remote_bitrate_estimator/test/bwe_test.cc +++ b/modules/remote_bitrate_estimator/test/bwe_test.cc @@ -76,11 +76,11 @@ class TestedEstimator : public RemoteBitrateObserver { } } - int64_t step_ms = estimator_->TimeUntilNextProcess(); + int64_t step_ms = std::max(estimator_->TimeUntilNextProcess(), 0); while ((clock_.TimeInMilliseconds() + step_ms) < packet_time_ms) { clock_.AdvanceTimeMilliseconds(step_ms); estimator_->Process(); - step_ms = estimator_->TimeUntilNextProcess(); + step_ms = std::max(estimator_->TimeUntilNextProcess(), 0); } estimator_->IncomingPacket(packet_time_ms, packet.payload_size(), packet.header()); @@ -195,13 +195,13 @@ class PacketProcessorRunner { if (queue_.empty()) { return; } - Packets to_transfer; Packets::iterator it = queue_.begin(); for (; it != queue_.end(); ++it) { if (it->send_time_us() > end_of_batch_time_us) { break; } } + Packets to_transfer; to_transfer.splice(to_transfer.begin(), queue_, queue_.begin(), it); batch->merge(to_transfer); } diff --git a/modules/remote_bitrate_estimator/test/bwe_test_framework.cc b/modules/remote_bitrate_estimator/test/bwe_test_framework.cc index b3cd7db9..194db4d3 100644 --- a/modules/remote_bitrate_estimator/test/bwe_test_framework.cc +++ b/modules/remote_bitrate_estimator/test/bwe_test_framework.cc @@ -154,6 +154,11 @@ void Packet::set_send_time_us(int64_t send_time_us) { send_time_us_ = send_time_us; } +void Packet::SetAbsSendTimeMs(int64_t abs_send_time_ms) { + header_.extension.absoluteSendTime = ((static_cast<int64_t>(abs_send_time_ms * + (1 << 18)) + 500) / 1000) & 0x00fffffful; +} + bool IsTimeSorted(const Packets& packets) { PacketsConstIt last_it = packets.begin(); for (PacketsConstIt it = last_it; it != packets.end(); ++it) { @@ -566,12 +571,8 @@ void VideoSender::RunFor(int64_t time_ms, Packets* in_out) { now_ms_ += time_ms; Packets new_packets; while (now_ms_ >= next_frame_ms_) { - prototype_header_.sequenceNumber++; prototype_header_.timestamp = kTimestampBase + static_cast<uint32_t>(next_frame_ms_ * 90.0); - prototype_header_.extension.absoluteSendTime = (kTimestampBase + - ((static_cast<int64_t>(next_frame_ms_ * (1 << 18)) + 500) / 1000)) & - 0x00fffffful; prototype_header_.extension.transmissionTimeOffset = 0; // Generate new packets for this frame, all with the same timestamp, @@ -581,9 +582,11 @@ void VideoSender::RunFor(int64_t time_ms, Packets* in_out) { int64_t send_time_us = next_frame_ms_ * 1000.0; uint32_t payload_size = frame_size_bytes_; while (payload_size > 0) { + ++prototype_header_.sequenceNumber; uint32_t size = std::min(kMaxPayloadSizeBytes, payload_size); new_packets.push_back(Packet(flow_ids()[0], send_time_us, size, prototype_header_)); + new_packets.back().SetAbsSendTimeMs(next_frame_ms_); payload_size -= size; } @@ -604,6 +607,98 @@ void AdaptiveVideoSender::GiveFeedback(const PacketSender::Feedback& feedback) { bytes_per_second_ = feedback.estimated_bps / 8; frame_size_bytes_ = (bytes_per_second_ * frame_period_ms_ + 500) / 1000; } + +PacedVideoSender::PacedVideoSender(PacketProcessorListener* listener, + uint32_t kbps, + AdaptiveVideoSender* source) + // It is important that the first_frame_offset and the initial time of + // clock_ are both zero, otherwise we can't have absolute time in this + // class. + : PacketSender(listener, source->flow_ids()), + clock_(0), + start_of_run_ms_(0), + pacer_(&clock_, this, PacedSender::kDefaultPaceMultiplier * kbps, 0), + source_(source) {} + +void PacedVideoSender::RunFor(int64_t time_ms, Packets* in_out) { + start_of_run_ms_ = clock_.TimeInMilliseconds(); + Packets generated_packets; + source_->RunFor(time_ms, &generated_packets); + Packets::iterator it = generated_packets.begin(); + // Run process periodically to allow the packets to be paced out. + const int kProcessIntervalMs = 10; + for (int64_t current_time = 0; current_time < time_ms; + current_time += kProcessIntervalMs) { + int64_t end_of_interval_us = + 1000 * (clock_.TimeInMilliseconds() + kProcessIntervalMs); + while (it != generated_packets.end() && + end_of_interval_us >= it->send_time_us()) { + // Time to send next packet to pacer. + pacer_.SendPacket(PacedSender::kNormalPriority, + it->header().ssrc, + it->header().sequenceNumber, + (it->send_time_us() + 500) / 1000, + it->payload_size(), + false); + pacer_queue_.push_back(*it); + const size_t kMaxPacerQueueSize = 1000; + if (pacer_queue_.size() > kMaxPacerQueueSize) { + pacer_queue_.pop_front(); + } + ++it; + } + clock_.AdvanceTimeMilliseconds(kProcessIntervalMs); + pacer_.Process(); + } + QueuePackets(in_out, (start_of_run_ms_ + time_ms) * 1000); +} + +void PacedVideoSender::QueuePackets(Packets* batch, + int64_t end_of_batch_time_us) { + queue_.merge(*batch); + if (queue_.empty()) { + return; + } + Packets::iterator it = queue_.begin(); + for (; it != queue_.end(); ++it) { + if (it->send_time_us() > end_of_batch_time_us) { + break; + } + } + Packets to_transfer; + to_transfer.splice(to_transfer.begin(), queue_, queue_.begin(), it); + batch->merge(to_transfer); +} + +void PacedVideoSender::GiveFeedback(const PacketSender::Feedback& feedback) { + source_->GiveFeedback(feedback); + pacer_.UpdateBitrate( + PacedSender::kDefaultPaceMultiplier * feedback.estimated_bps / 1000, 0); +} + +bool PacedVideoSender::TimeToSendPacket(uint32_t ssrc, + uint16_t sequence_number, + int64_t capture_time_ms, + bool retransmission) { + for (Packets::iterator it = pacer_queue_.begin(); it != pacer_queue_.end(); + ++it) { + if (it->header().sequenceNumber == sequence_number) { + int64_t pace_out_time_ms = clock_.TimeInMilliseconds(); + // Make sure a packet is never paced out earlier than when it was put into + // the pacer. + assert(1000 * pace_out_time_ms >= it->send_time_us()); + it->SetAbsSendTimeMs(pace_out_time_ms); + it->set_send_time_us(1000 * pace_out_time_ms); + queue_.push_back(*it); + return true; + } + } + return false; +} + +int PacedVideoSender::TimeToSendPadding(int bytes) { + return 0; +} } // namespace bwe } // namespace testing } // namespace webrtc diff --git a/modules/remote_bitrate_estimator/test/bwe_test_framework.h b/modules/remote_bitrate_estimator/test/bwe_test_framework.h index 8af07f91..0ab3b5f9 100644 --- a/modules/remote_bitrate_estimator/test/bwe_test_framework.h +++ b/modules/remote_bitrate_estimator/test/bwe_test_framework.h @@ -22,7 +22,9 @@ #include <vector> #include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/pacing/include/paced_sender.h" #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h" +#include "webrtc/system_wrappers/interface/clock.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" namespace webrtc { @@ -161,6 +163,7 @@ class Packet { int64_t creation_time_us() const { return creation_time_us_; } void set_send_time_us(int64_t send_time_us); int64_t send_time_us() const { return send_time_us_; } + void SetAbsSendTimeMs(int64_t abs_send_time_ms); uint32_t payload_size() const { return payload_size_; } const RTPHeader& header() const { return header_; } @@ -391,9 +394,9 @@ class VideoSender : public PacketSender { uint32_t max_payload_size_bytes() const { return kMaxPayloadSizeBytes; } uint32_t bytes_per_second() const { return bytes_per_second_; } - virtual uint32_t GetCapacityKbps() const; + virtual uint32_t GetCapacityKbps() const OVERRIDE; - virtual void RunFor(int64_t time_ms, Packets* in_out); + virtual void RunFor(int64_t time_ms, Packets* in_out) OVERRIDE; protected: const uint32_t kMaxPayloadSizeBytes; @@ -417,12 +420,43 @@ class AdaptiveVideoSender : public VideoSender { float first_frame_offset); virtual ~AdaptiveVideoSender() {} - virtual int GetFeedbackIntervalMs() const { return 100; } - virtual void GiveFeedback(const Feedback& feedback); + virtual int GetFeedbackIntervalMs() const OVERRIDE { return 100; } + virtual void GiveFeedback(const Feedback& feedback) OVERRIDE; -private: + private: DISALLOW_IMPLICIT_CONSTRUCTORS(AdaptiveVideoSender); }; + +class PacedVideoSender : public PacketSender, public PacedSender::Callback { + public: + PacedVideoSender(PacketProcessorListener* listener, + uint32_t kbps, AdaptiveVideoSender* source); + virtual ~PacedVideoSender() {} + + virtual int GetFeedbackIntervalMs() const OVERRIDE { return 100; } + virtual void GiveFeedback(const Feedback& feedback) OVERRIDE; + virtual void RunFor(int64_t time_ms, Packets* in_out) OVERRIDE; + + // Implements PacedSender::Callback. + virtual bool TimeToSendPacket(uint32_t ssrc, + uint16_t sequence_number, + int64_t capture_time_ms, + bool retransmission) OVERRIDE; + virtual int TimeToSendPadding(int bytes) OVERRIDE; + + private: + void QueuePackets(Packets* batch, int64_t end_of_batch_time_us); + + static const int64_t kInitialTimeMs = 0; + SimulatedClock clock_; + int64_t start_of_run_ms_; + PacedSender pacer_; + Packets pacer_queue_; + Packets queue_; + AdaptiveVideoSender* source_; + + DISALLOW_IMPLICIT_CONSTRUCTORS(PacedVideoSender); +}; } // namespace bwe } // namespace testing } // namespace webrtc diff --git a/modules/remote_bitrate_estimator/test/bwe_test_framework_unittest.cc b/modules/remote_bitrate_estimator/test/bwe_test_framework_unittest.cc index ec329268..6e735785 100644 --- a/modules/remote_bitrate_estimator/test/bwe_test_framework_unittest.cc +++ b/modules/remote_bitrate_estimator/test/bwe_test_framework_unittest.cc @@ -500,7 +500,7 @@ TEST(BweTestFramework_JitterFilterTest, Jitter1031) { TestJitterFilter(1031); } -static void TestReorderFilter(uint32_t reorder_percent, uint32_t near) { +static void TestReorderFilter(uint32_t reorder_percent, uint32_t near_value) { const uint32_t kPacketCount = 10000; // Generate packets with 10 ms interval. @@ -533,7 +533,8 @@ static void TestReorderFilter(uint32_t reorder_percent, uint32_t near) { // Because reordering is random, we allow a threshold when comparing. The // maximum distance a packet can be moved is PacketCount - 1. - EXPECT_NEAR(((kPacketCount - 1) * reorder_percent) / 100, distance, near); + EXPECT_NEAR( + ((kPacketCount - 1) * reorder_percent) / 100, distance, near_value); } TEST(BweTestFramework_ReorderFilterTest, Reorder0) { diff --git a/modules/rtp_rtcp/interface/rtp_header_parser.h b/modules/rtp_rtcp/interface/rtp_header_parser.h index a13f5b80..2809996b 100644 --- a/modules/rtp_rtcp/interface/rtp_header_parser.h +++ b/modules/rtp_rtcp/interface/rtp_header_parser.h @@ -23,13 +23,14 @@ class RtpHeaderParser { virtual ~RtpHeaderParser() {} // Returns true if the packet is an RTCP packet, false otherwise. - static bool IsRtcp(const uint8_t* packet, int length); + static bool IsRtcp(const uint8_t* packet, size_t length); // Parses the packet and stores the parsed packet in |header|. Returns true on // success, false otherwise. // This method is thread-safe in the sense that it can parse multiple packets // at once. - virtual bool Parse(const uint8_t* packet, int length, + virtual bool Parse(const uint8_t* packet, + size_t length, RTPHeader* header) const = 0; // Registers an RTP header extension and binds it to |id|. diff --git a/modules/rtp_rtcp/interface/rtp_payload_registry.h b/modules/rtp_rtcp/interface/rtp_payload_registry.h index 965f4b02..327ea165 100644 --- a/modules/rtp_rtcp/interface/rtp_payload_registry.h +++ b/modules/rtp_rtcp/interface/rtp_payload_registry.h @@ -25,17 +25,15 @@ class RTPPayloadStrategy { virtual bool CodecsMustBeUnique() const = 0; - virtual bool PayloadIsCompatible( - const ModuleRTPUtility::Payload& payload, - const uint32_t frequency, - const uint8_t channels, - const uint32_t rate) const = 0; + virtual bool PayloadIsCompatible(const RtpUtility::Payload& payload, + const uint32_t frequency, + const uint8_t channels, + const uint32_t rate) const = 0; - virtual void UpdatePayloadRate( - ModuleRTPUtility::Payload* payload, - const uint32_t rate) const = 0; + virtual void UpdatePayloadRate(RtpUtility::Payload* payload, + const uint32_t rate) const = 0; - virtual ModuleRTPUtility::Payload* CreatePayloadType( + virtual RtpUtility::Payload* CreatePayloadType( const char payloadName[RTP_PAYLOAD_NAME_SIZE], const int8_t payloadType, const uint32_t frequency, @@ -43,7 +41,7 @@ class RTPPayloadStrategy { const uint32_t rate) const = 0; virtual int GetPayloadTypeFrequency( - const ModuleRTPUtility::Payload& payload) const = 0; + const RtpUtility::Payload& payload) const = 0; static RTPPayloadStrategy* CreateStrategy(const bool handling_audio); @@ -99,9 +97,8 @@ class RTPPayloadRegistry { int GetPayloadTypeFrequency(uint8_t payload_type) const; - bool PayloadTypeToPayload( - const uint8_t payload_type, - ModuleRTPUtility::Payload*& payload) const; + bool PayloadTypeToPayload(const uint8_t payload_type, + RtpUtility::Payload*& payload) const; void ResetLastReceivedPayloadTypes() { CriticalSectionScoped cs(crit_sect_.get()); @@ -151,7 +148,7 @@ class RTPPayloadRegistry { bool IsRtxInternal(const RTPHeader& header) const; scoped_ptr<CriticalSectionWrapper> crit_sect_; - ModuleRTPUtility::PayloadTypeMap payload_type_map_; + RtpUtility::PayloadTypeMap payload_type_map_; scoped_ptr<RTPPayloadStrategy> rtp_payload_strategy_; int8_t red_payload_type_; int8_t ulpfec_payload_type_; diff --git a/modules/rtp_rtcp/interface/rtp_rtcp.h b/modules/rtp_rtcp/interface/rtp_rtcp.h index 95c565f0..235ca849 100644 --- a/modules/rtp_rtcp/interface/rtp_rtcp.h +++ b/modules/rtp_rtcp/interface/rtp_rtcp.h @@ -67,6 +67,7 @@ class RtpRtcp : public Module { RtpAudioFeedback* audio_messages; RemoteBitrateEstimator* remote_bitrate_estimator; PacedSender* paced_sender; + BitrateStatisticsObserver* send_bitrate_observer; }; /* @@ -203,6 +204,10 @@ class RtpRtcp : public Module { */ virtual int32_t SetSequenceNumber(const uint16_t seq) = 0; + virtual void SetRtpStateForSsrc(uint32_t ssrc, + const RtpState& rtp_state) = 0; + virtual bool GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) = 0; + /* * Get SSRC */ @@ -305,13 +310,6 @@ class RtpRtcp : public Module { uint32_t* nackRate) const = 0; /* - * Called on any new send bitrate estimate. - */ - virtual void RegisterVideoBitrateObserver( - BitrateStatisticsObserver* observer) = 0; - virtual BitrateStatisticsObserver* GetVideoBitrateObserver() const = 0; - - /* * Used by the codec module to deliver a video or audio frame for * packetization. * diff --git a/modules/rtp_rtcp/interface/rtp_rtcp_defines.h b/modules/rtp_rtcp/interface/rtp_rtcp_defines.h index 6f99f938..e1bec5fc 100644 --- a/modules/rtp_rtcp/interface/rtp_rtcp_defines.h +++ b/modules/rtp_rtcp/interface/rtp_rtcp_defines.h @@ -192,6 +192,20 @@ struct RtcpReceiveTimeInfo { typedef std::list<RTCPReportBlock> ReportBlockList; +struct RtpState { + RtpState() + : sequence_number(0), + start_timestamp(0), + timestamp(0), + capture_time_ms(-1), + last_timestamp_time_ms(-1) {} + uint16_t sequence_number; + uint32_t start_timestamp; + uint32_t timestamp; + int64_t capture_time_ms; + int64_t last_timestamp_time_ms; +}; + class RtpData { public: diff --git a/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h index 03156c79..fe20c6a3 100644 --- a/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h +++ b/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h @@ -74,6 +74,9 @@ class MockRtpRtcp : public RtpRtcp { uint16_t()); MOCK_METHOD1(SetSequenceNumber, int32_t(const uint16_t seq)); + MOCK_METHOD2(SetRtpStateForSsrc, + void(uint32_t ssrc, const RtpState& rtp_state)); + MOCK_METHOD2(GetRtpStateForSsrc, bool(uint32_t ssrc, RtpState* rtp_state)); MOCK_CONST_METHOD0(SSRC, uint32_t()); MOCK_METHOD1(SetSSRC, diff --git a/modules/rtp_rtcp/source/H264/rtp_sender_h264.cc b/modules/rtp_rtcp/source/H264/rtp_sender_h264.cc index d62f50b4..6560209c 100644 --- a/modules/rtp_rtcp/source/H264/rtp_sender_h264.cc +++ b/modules/rtp_rtcp/source/H264/rtp_sender_h264.cc @@ -238,9 +238,14 @@ RTPSenderH264::SendH264FillerData(const WebRtcRTPHeader* rtpHeader, dataBuffer[0] = static_cast<uint8_t>(0x80); // version 2 dataBuffer[1] = rtpHeader->header.payloadType; - ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+2, _rtpSender.IncrementSequenceNumber()); // get the current SequenceNumber and add by 1 after returning - ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+4, rtpHeader->header.timestamp); - ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+8, rtpHeader->header.ssrc); + RtpUtility::AssignUWord16ToBuffer( + dataBuffer + 2, + _rtpSender.IncrementSequenceNumber()); // get the current + // SequenceNumber and add by 1 + // after returning + RtpUtility::AssignUWord32ToBuffer(dataBuffer + 4, + rtpHeader->header.timestamp); + RtpUtility::AssignUWord32ToBuffer(dataBuffer + 8, rtpHeader->header.ssrc); // set filler NALU type dataBuffer[12] = 12; // NRI field = 0, type 12 @@ -361,8 +366,12 @@ RTPSenderH264::SendH264SVCRelayPacket(const WebRtcRTPHeader* rtpHeader, // _sequenceNumber will not work for re-ordering by NACK from original sender // engine responsible for this - ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+2, _rtpSender.IncrementSequenceNumber()); // get the current SequenceNumber and add by 1 after returning - //ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+8, ssrc); + RtpUtility::AssignUWord16ToBuffer( + dataBuffer + 2, + _rtpSender.IncrementSequenceNumber()); // get the current + // SequenceNumber and add by 1 + // after returning + // RtpUtility::AssignUWord32ToBuffer(dataBuffer+8, ssrc); // how do we know it's the last relayed packet in a frame? // 1) packets arrive in order, the engine manages that diff --git a/modules/rtp_rtcp/source/fec_receiver_impl.cc b/modules/rtp_rtcp/source/fec_receiver_impl.cc index 0d6c174a..e795841f 100644 --- a/modules/rtp_rtcp/source/fec_receiver_impl.cc +++ b/modules/rtp_rtcp/source/fec_receiver_impl.cc @@ -171,7 +171,7 @@ int32_t FecReceiverImpl::AddReceivedRedPacket( payload_data_length - REDHeaderLength); received_packet->pkt->length = payload_data_length - REDHeaderLength; received_packet->ssrc = - ModuleRTPUtility::BufferToUWord32(&incoming_rtp_packet[8]); + RtpUtility::BufferToUWord32(&incoming_rtp_packet[8]); } else { // copy the RTP header diff --git a/modules/rtp_rtcp/source/fec_test_helper.cc b/modules/rtp_rtcp/source/fec_test_helper.cc index 176954f9..0ffd5bf4 100644 --- a/modules/rtp_rtcp/source/fec_test_helper.cc +++ b/modules/rtp_rtcp/source/fec_test_helper.cc @@ -86,9 +86,9 @@ void FrameGenerator::BuildRtpHeader(uint8_t* data, const RTPHeader* header) { data[0] = 0x80; // Version 2. data[1] = header->payloadType; data[1] |= (header->markerBit ? kRtpMarkerBitMask : 0); - ModuleRTPUtility::AssignUWord16ToBuffer(data + 2, header->sequenceNumber); - ModuleRTPUtility::AssignUWord32ToBuffer(data + 4, header->timestamp); - ModuleRTPUtility::AssignUWord32ToBuffer(data + 8, header->ssrc); + RtpUtility::AssignUWord16ToBuffer(data + 2, header->sequenceNumber); + RtpUtility::AssignUWord32ToBuffer(data + 4, header->timestamp); + RtpUtility::AssignUWord32ToBuffer(data + 8, header->ssrc); } } // namespace webrtc diff --git a/modules/rtp_rtcp/source/forward_error_correction.cc b/modules/rtp_rtcp/source/forward_error_correction.cc index 31303c8a..b02ea086 100644 --- a/modules/rtp_rtcp/source/forward_error_correction.cc +++ b/modules/rtp_rtcp/source/forward_error_correction.cc @@ -230,7 +230,7 @@ void ForwardErrorCorrection::GenerateFecBitStrings( Packet* media_packet = *media_list_it; // Assign network-ordered media payload length. - ModuleRTPUtility::AssignUWord16ToBuffer( + RtpUtility::AssignUWord16ToBuffer( media_payload_length, media_packet->length - kRtpHeaderSize); fec_packet_length = media_packet->length + fec_rtp_offset; @@ -432,7 +432,7 @@ void ForwardErrorCorrection::GenerateFecUlpHeaders( // -- ULP header -- // Copy the payload size to the protection length field. // (We protect the entire packet.) - ModuleRTPUtility::AssignUWord16ToBuffer( + RtpUtility::AssignUWord16ToBuffer( &generated_fec_packets_[i].data[10], generated_fec_packets_[i].length - kFecHeaderSize - ulp_header_size); @@ -537,7 +537,7 @@ void ForwardErrorCorrection::InsertFECPacket( fec_packet->ssrc = rx_packet->ssrc; const uint16_t seq_num_base = - ModuleRTPUtility::BufferToUWord16(&fec_packet->pkt->data[2]); + RtpUtility::BufferToUWord16(&fec_packet->pkt->data[2]); const uint16_t maskSizeBytes = (fec_packet->pkt->data[0] & 0x40) ? kMaskSizeLBitSet : kMaskSizeLBitClear; // L bit set? @@ -603,6 +603,23 @@ void ForwardErrorCorrection::InsertPackets( while (!received_packet_list->empty()) { ReceivedPacket* rx_packet = received_packet_list->front(); + // Check for discarding oldest FEC packet, to avoid wrong FEC decoding from + // sequence number wrap-around. Detection of old FEC packet is based on + // sequence number difference of received packet and oldest packet in FEC + // packet list. + // TODO(marpan/holmer): We should be able to improve detection/discarding of + // old FEC packets based on timestamp information or better sequence number + // thresholding (e.g., to distinguish between wrap-around and reordering). + if (!fec_packet_list_.empty()) { + uint16_t seq_num_diff = abs( + static_cast<int>(rx_packet->seq_num) - + static_cast<int>(fec_packet_list_.front()->seq_num)); + if (seq_num_diff > 0x3fff) { + DiscardFECPacket(fec_packet_list_.front()); + fec_packet_list_.pop_front(); + } + } + if (rx_packet->is_fec) { InsertFECPacket(rx_packet, recovered_packet_list); } else { @@ -633,7 +650,7 @@ void ForwardErrorCorrection::InitRecovery(const FecPacket* fec_packet, // Copy FEC payload, skipping the ULP header. memcpy(&recovered->pkt->data[kRtpHeaderSize], &fec_packet->pkt->data[kFecHeaderSize + ulp_header_size], - ModuleRTPUtility::BufferToUWord16(protection_length)); + RtpUtility::BufferToUWord16(protection_length)); // Copy the length recovery field. memcpy(recovered->length_recovery, &fec_packet->pkt->data[8], 2); // Copy the first 2 bytes of the FEC header. @@ -641,8 +658,7 @@ void ForwardErrorCorrection::InitRecovery(const FecPacket* fec_packet, // Copy the 5th to 8th bytes of the FEC header. memcpy(&recovered->pkt->data[4], &fec_packet->pkt->data[4], 4); // Set the SSRC field. - ModuleRTPUtility::AssignUWord32ToBuffer(&recovered->pkt->data[8], - fec_packet->ssrc); + RtpUtility::AssignUWord32ToBuffer(&recovered->pkt->data[8], fec_packet->ssrc); } void ForwardErrorCorrection::FinishRecovery(RecoveredPacket* recovered) { @@ -651,12 +667,11 @@ void ForwardErrorCorrection::FinishRecovery(RecoveredPacket* recovered) { recovered->pkt->data[0] &= 0xbf; // Clear the 2nd bit. // Set the SN field. - ModuleRTPUtility::AssignUWord16ToBuffer(&recovered->pkt->data[2], - recovered->seq_num); + RtpUtility::AssignUWord16ToBuffer(&recovered->pkt->data[2], + recovered->seq_num); // Recover the packet length. recovered->pkt->length = - ModuleRTPUtility::BufferToUWord16(recovered->length_recovery) + - kRtpHeaderSize; + RtpUtility::BufferToUWord16(recovered->length_recovery) + kRtpHeaderSize; } void ForwardErrorCorrection::XorPackets(const Packet* src_packet, @@ -671,8 +686,8 @@ void ForwardErrorCorrection::XorPackets(const Packet* src_packet, } // XOR with the network-ordered payload size. uint8_t media_payload_length[2]; - ModuleRTPUtility::AssignUWord16ToBuffer(media_payload_length, - src_packet->length - kRtpHeaderSize); + RtpUtility::AssignUWord16ToBuffer(media_payload_length, + src_packet->length - kRtpHeaderSize); dst_packet->length_recovery[0] ^= media_payload_length[0]; dst_packet->length_recovery[1] ^= media_payload_length[1]; diff --git a/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h b/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h index ccf82e5d..f577cbaa 100644 --- a/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h +++ b/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h @@ -21,21 +21,21 @@ class MockRTPPayloadStrategy : public RTPPayloadStrategy { MOCK_CONST_METHOD0(CodecsMustBeUnique, bool()); MOCK_CONST_METHOD4(PayloadIsCompatible, - bool(const ModuleRTPUtility::Payload& payload, - const uint32_t frequency, - const uint8_t channels, - const uint32_t rate)); + bool(const RtpUtility::Payload& payload, + const uint32_t frequency, + const uint8_t channels, + const uint32_t rate)); MOCK_CONST_METHOD2(UpdatePayloadRate, - void(ModuleRTPUtility::Payload* payload, const uint32_t rate)); - MOCK_CONST_METHOD1(GetPayloadTypeFrequency, int( - const ModuleRTPUtility::Payload& payload)); - MOCK_CONST_METHOD5(CreatePayloadType, - ModuleRTPUtility::Payload*( - const char payloadName[RTP_PAYLOAD_NAME_SIZE], - const int8_t payloadType, - const uint32_t frequency, - const uint8_t channels, - const uint32_t rate)); + void(RtpUtility::Payload* payload, const uint32_t rate)); + MOCK_CONST_METHOD1(GetPayloadTypeFrequency, + int(const RtpUtility::Payload& payload)); + MOCK_CONST_METHOD5( + CreatePayloadType, + RtpUtility::Payload*(const char payloadName[RTP_PAYLOAD_NAME_SIZE], + const int8_t payloadType, + const uint32_t frequency, + const uint8_t channels, + const uint32_t rate)); }; } // namespace webrtc diff --git a/modules/rtp_rtcp/source/producer_fec.cc b/modules/rtp_rtcp/source/producer_fec.cc index 3173d3d5..747cd89e 100644 --- a/modules/rtp_rtcp/source/producer_fec.cc +++ b/modules/rtp_rtcp/source/producer_fec.cc @@ -61,7 +61,7 @@ void RedPacket::CreateHeader(const uint8_t* rtp_header, int header_length, void RedPacket::SetSeqNum(int seq_num) { assert(seq_num >= 0 && seq_num < (1<<16)); - ModuleRTPUtility::AssignUWord16ToBuffer(&data_[2], seq_num); + RtpUtility::AssignUWord16ToBuffer(&data_[2], seq_num); } void RedPacket::AssignPayload(const uint8_t* payload, int length) { diff --git a/modules/rtp_rtcp/source/receive_statistics_impl.cc b/modules/rtp_rtcp/source/receive_statistics_impl.cc index aa7c9c57..e3bc95f7 100644 --- a/modules/rtp_rtcp/source/receive_statistics_impl.cc +++ b/modules/rtp_rtcp/source/receive_statistics_impl.cc @@ -133,11 +133,12 @@ void StreamStatisticianImpl::UpdateCounters(const RTPHeader& header, void StreamStatisticianImpl::UpdateJitter(const RTPHeader& header, uint32_t receive_time_secs, uint32_t receive_time_frac) { - uint32_t receive_time_rtp = ModuleRTPUtility::ConvertNTPTimeToRTP( - receive_time_secs, receive_time_frac, header.payload_type_frequency); - uint32_t last_receive_time_rtp = ModuleRTPUtility::ConvertNTPTimeToRTP( - last_receive_time_secs_, last_receive_time_frac_, - header.payload_type_frequency); + uint32_t receive_time_rtp = RtpUtility::ConvertNTPTimeToRTP( + receive_time_secs, receive_time_frac, header.payload_type_frequency); + uint32_t last_receive_time_rtp = + RtpUtility::ConvertNTPTimeToRTP(last_receive_time_secs_, + last_receive_time_frac_, + header.payload_type_frequency); int32_t time_diff_samples = (receive_time_rtp - last_receive_time_rtp) - (header.timestamp - last_received_timestamp_); diff --git a/modules/rtp_rtcp/source/rtcp_packet.cc b/modules/rtp_rtcp/source/rtcp_packet.cc index f6d3bd3d..68da3aeb 100644 --- a/modules/rtp_rtcp/source/rtcp_packet.cc +++ b/modules/rtp_rtcp/source/rtcp_packet.cc @@ -61,15 +61,15 @@ void AssignUWord8(uint8_t* buffer, size_t* offset, uint8_t value) { buffer[(*offset)++] = value; } void AssignUWord16(uint8_t* buffer, size_t* offset, uint16_t value) { - ModuleRTPUtility::AssignUWord16ToBuffer(buffer + *offset, value); + RtpUtility::AssignUWord16ToBuffer(buffer + *offset, value); *offset += 2; } void AssignUWord24(uint8_t* buffer, size_t* offset, uint32_t value) { - ModuleRTPUtility::AssignUWord24ToBuffer(buffer + *offset, value); + RtpUtility::AssignUWord24ToBuffer(buffer + *offset, value); *offset += 3; } void AssignUWord32(uint8_t* buffer, size_t* offset, uint32_t value) { - ModuleRTPUtility::AssignUWord32ToBuffer(buffer + *offset, value); + RtpUtility::AssignUWord32ToBuffer(buffer + *offset, value); *offset += 4; } diff --git a/modules/rtp_rtcp/source/rtcp_receiver.cc b/modules/rtp_rtcp/source/rtcp_receiver.cc index 896bd5f4..b38ae1f0 100644 --- a/modules/rtp_rtcp/source/rtcp_receiver.cc +++ b/modules/rtp_rtcp/source/rtcp_receiver.cc @@ -1361,8 +1361,6 @@ int32_t RTCPReceiver::UpdateTMMBR() { void RTCPReceiver::RegisterRtcpStatisticsCallback( RtcpStatisticsCallback* callback) { CriticalSectionScoped cs(_criticalSectionFeedbacks); - if (callback != NULL) - assert(stats_callback_ == NULL); stats_callback_ = callback; } diff --git a/modules/rtp_rtcp/source/rtcp_sender.cc b/modules/rtp_rtcp/source/rtcp_sender.cc index d73de9c4..b9ab0c1e 100644 --- a/modules/rtp_rtcp/source/rtcp_sender.cc +++ b/modules/rtp_rtcp/source/rtcp_sender.cc @@ -708,24 +708,24 @@ int32_t RTCPSender::BuildSR(const FeedbackState& feedback_state, pos++; // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; // NTP - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, NTPsec); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, NTPsec); pos += 4; - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, NTPfrac); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, NTPfrac); pos += 4; - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, RTPtime); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, RTPtime); pos += 4; //sender's packet count - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, - feedback_state.packet_count_sent); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, + feedback_state.packet_count_sent); pos += 4; //sender's octet count - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, - feedback_state.byte_count_sent); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, + feedback_state.byte_count_sent); pos += 4; uint8_t numberOfReportBlocks = 0; @@ -741,7 +741,7 @@ int32_t RTCPSender::BuildSR(const FeedbackState& feedback_state, rtcpbuffer[posNumberOfReportBlocks] += numberOfReportBlocks; uint16_t len = uint16_t((pos/4) -1); - ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer+2, len); + RtpUtility::AssignUWord16ToBuffer(rtcpbuffer + 2, len); return 0; } @@ -767,7 +767,7 @@ int32_t RTCPSender::BuildSDEC(uint8_t* rtcpbuffer, int& pos) { pos++; // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; // CNAME = 1 @@ -802,7 +802,7 @@ int32_t RTCPSender::BuildSDEC(uint8_t* rtcpbuffer, int& pos) { uint32_t SSRC = it->first; // Add SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, SSRC); pos += 4; // CNAME = 1 @@ -833,8 +833,7 @@ int32_t RTCPSender::BuildSDEC(uint8_t* rtcpbuffer, int& pos) { } // in 32-bit words minus one and we don't count the header uint16_t buffer_length = (SDESLength / 4) - 1; - ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer + SDESLengthPos, - buffer_length); + RtpUtility::AssignUWord16ToBuffer(rtcpbuffer + SDESLengthPos, buffer_length); return 0; } @@ -859,7 +858,7 @@ RTCPSender::BuildRR(uint8_t* rtcpbuffer, pos++; // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; uint8_t numberOfReportBlocks = 0; @@ -874,7 +873,7 @@ RTCPSender::BuildRR(uint8_t* rtcpbuffer, rtcpbuffer[posNumberOfReportBlocks] += numberOfReportBlocks; uint16_t len = uint16_t((pos)/4 -1); - ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer+2, len); + RtpUtility::AssignUWord16ToBuffer(rtcpbuffer + 2, len); return 0; } @@ -925,8 +924,8 @@ RTCPSender::BuildExtendedJitterReport( rtcpbuffer[pos++]=(uint8_t)(1); // Add inter-arrival jitter - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, - jitterTransmissionTimeOffset); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, + jitterTransmissionTimeOffset); pos += 4; return 0; } @@ -949,11 +948,11 @@ RTCPSender::BuildPLI(uint8_t* rtcpbuffer, int& pos) rtcpbuffer[pos++]=(uint8_t)(2); // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; // Add the remote SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _remoteSSRC); pos += 4; return 0; } @@ -979,7 +978,7 @@ int32_t RTCPSender::BuildFIR(uint8_t* rtcpbuffer, rtcpbuffer[pos++] = (uint8_t)(4); // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; // RFC 5104 4.3.1.2. Semantics @@ -990,7 +989,7 @@ int32_t RTCPSender::BuildFIR(uint8_t* rtcpbuffer, rtcpbuffer[pos++] = (uint8_t)0; // Additional Feedback Control Information (FCI) - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _remoteSSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _remoteSSRC); pos += 4; rtcpbuffer[pos++] = (uint8_t)(_sequenceNumberFIR); @@ -1025,18 +1024,18 @@ RTCPSender::BuildSLI(uint8_t* rtcpbuffer, int& pos, const uint8_t pictureID) rtcpbuffer[pos++]=(uint8_t)(3); // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; // Add the remote SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _remoteSSRC); pos += 4; // Add first, number & picture ID 6 bits // first = 0, 13 - bits // number = 0x1fff, 13 - bits only ones for now uint32_t sliField = (0x1fff << 6)+ (0x3f & pictureID); - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, sliField); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, sliField); pos += 4; return 0; } @@ -1090,11 +1089,11 @@ RTCPSender::BuildRPSI(uint8_t* rtcpbuffer, rtcpbuffer[pos++]=size; // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; // Add the remote SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _remoteSSRC); pos += 4; // calc padding length @@ -1147,11 +1146,11 @@ RTCPSender::BuildREMB(uint8_t* rtcpbuffer, int& pos) rtcpbuffer[pos++]=_lengthRembSSRC + 4; // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; // Remote SSRC must be 0 - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, 0); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, 0); pos += 4; rtcpbuffer[pos++]='R'; @@ -1177,8 +1176,8 @@ RTCPSender::BuildREMB(uint8_t* rtcpbuffer, int& pos) rtcpbuffer[pos++]=(uint8_t)(brMantissa); for (int i = 0; i < _lengthRembSSRC; i++) - { - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _rembSSRC[i]); + { + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _rembSSRC[i]); pos += 4; } return 0; @@ -1264,7 +1263,7 @@ int32_t RTCPSender::BuildTMMBR(ModuleRtpRtcpImpl* rtp_rtcp_module, rtcpbuffer[pos++]=(uint8_t)(4); // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; // RFC 5104 4.2.1.2. Semantics @@ -1276,7 +1275,7 @@ int32_t RTCPSender::BuildTMMBR(ModuleRtpRtcpImpl* rtp_rtcp_module, rtcpbuffer[pos++]=(uint8_t)0; // Additional Feedback Control Information (FCI) - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _remoteSSRC); pos += 4; uint32_t bitRate = _tmmbr_Send*1000; @@ -1324,7 +1323,7 @@ RTCPSender::BuildTMMBN(uint8_t* rtcpbuffer, int& pos) pos++; // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; // RFC 5104 4.2.2.2. Semantics @@ -1342,7 +1341,7 @@ RTCPSender::BuildTMMBN(uint8_t* rtcpbuffer, int& pos) if (boundingSet->Tmmbr(n) > 0) { uint32_t tmmbrSSRC = boundingSet->Ssrc(n); - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, tmmbrSSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, tmmbrSSRC); pos += 4; uint32_t bitRate = boundingSet->Tmmbr(n) * 1000; @@ -1395,11 +1394,11 @@ RTCPSender::BuildAPP(uint8_t* rtcpbuffer, int& pos) rtcpbuffer[pos++]=(uint8_t)(length); // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; // Add our application name - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _appName); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _appName); pos += 4; // Add the data @@ -1433,11 +1432,11 @@ RTCPSender::BuildNACK(uint8_t* rtcpbuffer, rtcpbuffer[pos++]=(uint8_t)(3); //setting it to one kNACK signal as default // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; // Add the remote SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _remoteSSRC); pos += 4; NACKStringBuilder stringBuilder; @@ -1464,9 +1463,9 @@ RTCPSender::BuildNACK(uint8_t* rtcpbuffer, } // Write the sequence number and the bitmask to the packet. assert(pos + 4 < IP_PACKET_SIZE); - ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer + pos, nack); + RtpUtility::AssignUWord16ToBuffer(rtcpbuffer + pos, nack); pos += 2; - ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer + pos, bitmask); + RtpUtility::AssignUWord16ToBuffer(rtcpbuffer + pos, bitmask); pos += 2; numOfNackFields++; } @@ -1497,13 +1496,13 @@ RTCPSender::BuildBYE(uint8_t* rtcpbuffer, int& pos) rtcpbuffer[pos++]=(uint8_t)(1 + _CSRCs); // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; // add CSRCs for(int i = 0; i < _CSRCs; i++) { - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _CSRC[i]); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _CSRC[i]); pos += 4; } } else @@ -1517,7 +1516,7 @@ RTCPSender::BuildBYE(uint8_t* rtcpbuffer, int& pos) rtcpbuffer[pos++]=(uint8_t)1; // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; } return 0; @@ -1546,7 +1545,7 @@ int32_t RTCPSender::BuildReceiverReferenceTime(uint8_t* buffer, buffer[pos++] = 4; // XR packet length. // Add our own SSRC. - ModuleRTPUtility::AssignUWord32ToBuffer(buffer + pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(buffer + pos, _SSRC); pos += 4; // 0 1 2 3 @@ -1566,9 +1565,9 @@ int32_t RTCPSender::BuildReceiverReferenceTime(uint8_t* buffer, buffer[pos++] = 2; // Block length. // NTP timestamp. - ModuleRTPUtility::AssignUWord32ToBuffer(buffer + pos, ntp_sec); + RtpUtility::AssignUWord32ToBuffer(buffer + pos, ntp_sec); pos += 4; - ModuleRTPUtility::AssignUWord32ToBuffer(buffer + pos, ntp_frac); + RtpUtility::AssignUWord32ToBuffer(buffer + pos, ntp_frac); pos += 4; return 0; @@ -1589,7 +1588,7 @@ int32_t RTCPSender::BuildDlrr(uint8_t* buffer, buffer[pos++] = 5; // XR packet length. // Add our own SSRC. - ModuleRTPUtility::AssignUWord32ToBuffer(buffer + pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(buffer + pos, _SSRC); pos += 4; // 0 1 2 3 @@ -1614,11 +1613,11 @@ int32_t RTCPSender::BuildDlrr(uint8_t* buffer, buffer[pos++] = 3; // Block length. // NTP timestamp. - ModuleRTPUtility::AssignUWord32ToBuffer(buffer + pos, info.sourceSSRC); + RtpUtility::AssignUWord32ToBuffer(buffer + pos, info.sourceSSRC); pos += 4; - ModuleRTPUtility::AssignUWord32ToBuffer(buffer + pos, info.lastRR); + RtpUtility::AssignUWord32ToBuffer(buffer + pos, info.lastRR); pos += 4; - ModuleRTPUtility::AssignUWord32ToBuffer(buffer + pos, info.delaySinceLastRR); + RtpUtility::AssignUWord32ToBuffer(buffer + pos, info.delaySinceLastRR); pos += 4; return 0; @@ -1644,7 +1643,7 @@ RTCPSender::BuildVoIPMetric(uint8_t* rtcpbuffer, int& pos) pos++; // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; // Add a VoIP metrics block @@ -1654,7 +1653,7 @@ RTCPSender::BuildVoIPMetric(uint8_t* rtcpbuffer, int& pos) rtcpbuffer[pos++]=8; // Add the remote SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _remoteSSRC); pos += 4; rtcpbuffer[pos++] = _xrVoIPMetric.lossRate; @@ -2210,33 +2209,33 @@ int32_t RTCPSender::WriteReportBlocksToBuffer( RTCPReportBlock* reportBlock = it->second; if (reportBlock) { // Remote SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+position, remoteSSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + position, remoteSSRC); position += 4; // fraction lost rtcpbuffer[position++] = reportBlock->fractionLost; // cumulative loss - ModuleRTPUtility::AssignUWord24ToBuffer(rtcpbuffer+position, - reportBlock->cumulativeLost); + RtpUtility::AssignUWord24ToBuffer(rtcpbuffer + position, + reportBlock->cumulativeLost); position += 3; // extended highest seq_no, contain the highest sequence number received - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+position, - reportBlock->extendedHighSeqNum); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + position, + reportBlock->extendedHighSeqNum); position += 4; // Jitter - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+position, - reportBlock->jitter); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + position, + reportBlock->jitter); position += 4; - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+position, - reportBlock->lastSR); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + position, + reportBlock->lastSR); position += 4; - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+position, - reportBlock->delaySinceLastSR); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + position, + reportBlock->delaySinceLastSR); position += 4; } } diff --git a/modules/rtp_rtcp/source/rtp_fec_unittest.cc b/modules/rtp_rtcp/source/rtp_fec_unittest.cc index fa847625..9d19fde1 100644 --- a/modules/rtp_rtcp/source/rtp_fec_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_fec_unittest.cc @@ -155,6 +155,174 @@ TEST_F(RtpFecTest, FecRecoveryWithLoss) { EXPECT_FALSE(IsRecoveryComplete()); } +// Verify that we don't use an old FEC packet for FEC decoding. +TEST_F(RtpFecTest, FecRecoveryWithSeqNumGapTwoFrames) { + const int kNumImportantPackets = 0; + const bool kUseUnequalProtection = false; + uint8_t kProtectionFactor = 20; + + // Two frames: first frame (old) with two media packets and 1 FEC packet. + // Second frame (new) with 3 media packets, and no FEC packets. + // ---Frame 1---- ----Frame 2------ + // #0(media) #1(media) #2(FEC) #65535(media) #0(media) #1(media). + // If we lose either packet 0 or 1 of second frame, FEC decoding should not + // try to decode using "old" FEC packet #2. + + // Construct media packets for first frame, starting at sequence number 0. + fec_seq_num_ = ConstructMediaPacketsSeqNum(2, 0); + + EXPECT_EQ(0, fec_->GenerateFEC(media_packet_list_, kProtectionFactor, + kNumImportantPackets, kUseUnequalProtection, + webrtc::kFecMaskBursty, &fec_packet_list_)); + // Expect 1 FEC packet. + EXPECT_EQ(1, static_cast<int>(fec_packet_list_.size())); + // Add FEC packet (seq#2) of this first frame to received list (i.e., assume + // the two media packet were lost). + memset(fec_loss_mask_, 0, sizeof(fec_loss_mask_)); + ReceivedPackets(fec_packet_list_, fec_loss_mask_, true); + + // Construct media packets for second frame, with sequence number wrap. + ClearList(&media_packet_list_); + fec_seq_num_ = ConstructMediaPacketsSeqNum(3, 65535); + + // Expect 3 media packets for this frame. + EXPECT_EQ(3, static_cast<int>(media_packet_list_.size())); + + // Second media packet lost (seq#0). + memset(media_loss_mask_, 0, sizeof(media_loss_mask_)); + media_loss_mask_[1] = 1; + // Add packets #65535, and #1 to received list. + ReceivedPackets(media_packet_list_, media_loss_mask_, false); + + EXPECT_EQ(0, + fec_->DecodeFEC(&received_packet_list_, &recovered_packet_list_)); + + // Expect that no decoding is done to get missing packet (seq#0) of second + // frame, using old FEC packet (seq#2) from first (old) frame. So number of + // recovered packets is 2, and not equal to number of media packets (=3). + EXPECT_EQ(2, static_cast<int>(recovered_packet_list_.size())); + EXPECT_TRUE(recovered_packet_list_.size() != media_packet_list_.size()); + FreeRecoveredPacketList(); +} + +// Verify we can still recovery frame if sequence number wrap occurs within +// the frame and FEC packet following wrap is received after media packets. +TEST_F(RtpFecTest, FecRecoveryWithSeqNumGapOneFrameRecovery) { + const int kNumImportantPackets = 0; + const bool kUseUnequalProtection = false; + uint8_t kProtectionFactor = 20; + + // One frame, with sequence number wrap in media packets. + // -----Frame 1---- + // #65534(media) #65535(media) #0(media) #1(FEC). + fec_seq_num_ = ConstructMediaPacketsSeqNum(3, 65534); + + EXPECT_EQ(0, fec_->GenerateFEC(media_packet_list_, kProtectionFactor, + kNumImportantPackets, kUseUnequalProtection, + webrtc::kFecMaskBursty, &fec_packet_list_)); + + // Expect 1 FEC packet. + EXPECT_EQ(1, static_cast<int>(fec_packet_list_.size())); + + // Lose one media packet (seq# 65535). + memset(media_loss_mask_, 0, sizeof(media_loss_mask_)); + memset(fec_loss_mask_, 0, sizeof(fec_loss_mask_)); + media_loss_mask_[1] = 1; + ReceivedPackets(media_packet_list_, media_loss_mask_, false); + // Add FEC packet to received list following the media packets. + ReceivedPackets(fec_packet_list_, fec_loss_mask_, true); + + EXPECT_EQ(0, + fec_->DecodeFEC(&received_packet_list_, &recovered_packet_list_)); + + // Expect 3 media packets in recovered list, and complete recovery. + // Wrap-around won't remove FEC packet, as it follows the wrap. + EXPECT_EQ(3, static_cast<int>(recovered_packet_list_.size())); + EXPECT_TRUE(IsRecoveryComplete()); + FreeRecoveredPacketList(); +} + +// Sequence number wrap occurs within the FEC packets for the frame. +// In this case we will discard FEC packet and full recovery is not expected. +// Same problem will occur if wrap is within media packets but FEC packet is +// received before the media packets. This may be improved if timing information +// is used to detect old FEC packets. +// TODO(marpan): Update test if wrap-around handling changes in FEC decoding. +TEST_F(RtpFecTest, FecRecoveryWithSeqNumGapOneFrameNoRecovery) { + const int kNumImportantPackets = 0; + const bool kUseUnequalProtection = false; + uint8_t kProtectionFactor = 200; + + // 1 frame: 3 media packets and 2 FEC packets. + // Sequence number wrap in FEC packets. + // -----Frame 1---- + // #65532(media) #65533(media) #65534(media) #65535(FEC) #0(FEC). + fec_seq_num_ = ConstructMediaPacketsSeqNum(3, 65532); + + EXPECT_EQ(0, fec_->GenerateFEC(media_packet_list_, kProtectionFactor, + kNumImportantPackets, kUseUnequalProtection, + webrtc::kFecMaskBursty, &fec_packet_list_)); + + // Expect 2 FEC packets. + EXPECT_EQ(2, static_cast<int>(fec_packet_list_.size())); + + // Lose the last two media packets (seq# 65533, 65534). + memset(media_loss_mask_, 0, sizeof(media_loss_mask_)); + memset(fec_loss_mask_, 0, sizeof(fec_loss_mask_)); + media_loss_mask_[1] = 1; + media_loss_mask_[2] = 1; + ReceivedPackets(media_packet_list_, media_loss_mask_, false); + ReceivedPackets(fec_packet_list_, fec_loss_mask_, true); + + EXPECT_EQ(0, + fec_->DecodeFEC(&received_packet_list_, &recovered_packet_list_)); + + // The two FEC packets are received and should allow for complete recovery, + // but because of the wrap the second FEC packet will be discarded, and only + // one media packet is recoverable. So exepct 2 media packets on recovered + // list and no complete recovery. + EXPECT_EQ(2, static_cast<int>(recovered_packet_list_.size())); + EXPECT_TRUE(recovered_packet_list_.size() != media_packet_list_.size()); + EXPECT_FALSE(IsRecoveryComplete()); + FreeRecoveredPacketList(); +} + +// Verify we can still recovery frame if FEC is received before media packets. +TEST_F(RtpFecTest, FecRecoveryWithFecOutOfOrder) { + const int kNumImportantPackets = 0; + const bool kUseUnequalProtection = false; + uint8_t kProtectionFactor = 20; + + // One frame: 3 media packets, 1 FEC packet. + // -----Frame 1---- + // #0(media) #1(media) #2(media) #3(FEC). + fec_seq_num_ = ConstructMediaPacketsSeqNum(3, 0); + + EXPECT_EQ(0, fec_->GenerateFEC(media_packet_list_, kProtectionFactor, + kNumImportantPackets, kUseUnequalProtection, + webrtc::kFecMaskBursty, &fec_packet_list_)); + + // Expect 1 FEC packet. + EXPECT_EQ(1, static_cast<int>(fec_packet_list_.size())); + + // Lose one media packet (seq# 1). + memset(media_loss_mask_, 0, sizeof(media_loss_mask_)); + memset(fec_loss_mask_, 0, sizeof(fec_loss_mask_)); + media_loss_mask_[1] = 1; + // Add FEC packet to received list before the media packets. + ReceivedPackets(fec_packet_list_, fec_loss_mask_, true); + // Add media packets to received list. + ReceivedPackets(media_packet_list_, media_loss_mask_, false); + + EXPECT_EQ(0, + fec_->DecodeFEC(&received_packet_list_, &recovered_packet_list_)); + + // Expect 3 media packets in recovered list, and complete recovery. + EXPECT_EQ(3, static_cast<int>(recovered_packet_list_.size())); + EXPECT_TRUE(IsRecoveryComplete()); + FreeRecoveredPacketList(); +} + // Test 50% protection with random mask type: Two cases are considered: // a 50% non-consecutive loss which can be fully recovered, and a 50% // consecutive loss which cannot be fully recovered. @@ -625,8 +793,6 @@ TEST_F(RtpFecTest, FecRecoveryNonConsecutivePacketsWrap) { EXPECT_FALSE(IsRecoveryComplete()); } -// TODO(marpan): Add more test cases. - void RtpFecTest::TearDown() { fec_->ResetState(&recovered_packet_list_); delete fec_; @@ -700,7 +866,7 @@ void RtpFecTest::ReceivedPackets(const PacketList& packet_list, int* loss_mask, // For media packets, the sequence number and marker bit is // obtained from RTP header. These were set in ConstructMediaPackets(). received_packet->seq_num = - webrtc::ModuleRTPUtility::BufferToUWord16(&packet->data[2]); + webrtc::RtpUtility::BufferToUWord16(&packet->data[2]); } else { // The sequence number, marker bit, and ssrc number are defined in the // RTP header of the FEC packet, which is not constructed in this test. @@ -755,12 +921,11 @@ int RtpFecTest::ConstructMediaPacketsSeqNum(int num_media_packets, // Only push one (fake) frame to the FEC. media_packet->data[1] &= 0x7f; - webrtc::ModuleRTPUtility::AssignUWord16ToBuffer(&media_packet->data[2], - sequence_number); - webrtc::ModuleRTPUtility::AssignUWord32ToBuffer(&media_packet->data[4], - time_stamp); - webrtc::ModuleRTPUtility::AssignUWord32ToBuffer(&media_packet->data[8], - ssrc_); + webrtc::RtpUtility::AssignUWord16ToBuffer(&media_packet->data[2], + sequence_number); + webrtc::RtpUtility::AssignUWord32ToBuffer(&media_packet->data[4], + time_stamp); + webrtc::RtpUtility::AssignUWord32ToBuffer(&media_packet->data[8], ssrc_); // Generate random values for payload. for (int j = 12; j < media_packet->length; ++j) { diff --git a/modules/rtp_rtcp/source/rtp_header_parser.cc b/modules/rtp_rtcp/source/rtp_header_parser.cc index bb24d4db..3fc26663 100644 --- a/modules/rtp_rtcp/source/rtp_header_parser.cc +++ b/modules/rtp_rtcp/source/rtp_header_parser.cc @@ -21,7 +21,8 @@ class RtpHeaderParserImpl : public RtpHeaderParser { RtpHeaderParserImpl(); virtual ~RtpHeaderParserImpl() {} - virtual bool Parse(const uint8_t* packet, int length, + virtual bool Parse(const uint8_t* packet, + size_t length, RTPHeader* header) const OVERRIDE; virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, @@ -31,7 +32,7 @@ class RtpHeaderParserImpl : public RtpHeaderParser { private: scoped_ptr<CriticalSectionWrapper> critical_section_; - RtpHeaderExtensionMap rtp_header_extension_map_; + RtpHeaderExtensionMap rtp_header_extension_map_ GUARDED_BY(critical_section_); }; RtpHeaderParser* RtpHeaderParser::Create() { @@ -41,14 +42,15 @@ RtpHeaderParser* RtpHeaderParser::Create() { RtpHeaderParserImpl::RtpHeaderParserImpl() : critical_section_(CriticalSectionWrapper::CreateCriticalSection()) {} -bool RtpHeaderParser::IsRtcp(const uint8_t* packet, int length) { - ModuleRTPUtility::RTPHeaderParser rtp_parser(packet, length); +bool RtpHeaderParser::IsRtcp(const uint8_t* packet, size_t length) { + RtpUtility::RtpHeaderParser rtp_parser(packet, length); return rtp_parser.RTCP(); } -bool RtpHeaderParserImpl::Parse(const uint8_t* packet, int length, - RTPHeader* header) const { - ModuleRTPUtility::RTPHeaderParser rtp_parser(packet, length); +bool RtpHeaderParserImpl::Parse(const uint8_t* packet, + size_t length, + RTPHeader* header) const { + RtpUtility::RtpHeaderParser rtp_parser(packet, length); memset(header, 0, sizeof(*header)); RtpHeaderExtensionMap map; diff --git a/modules/rtp_rtcp/source/rtp_payload_registry.cc b/modules/rtp_rtcp/source/rtp_payload_registry.cc index db2e4cd3..ec05a73b 100644 --- a/modules/rtp_rtcp/source/rtp_payload_registry.cc +++ b/modules/rtp_rtcp/source/rtp_payload_registry.cc @@ -29,7 +29,7 @@ RTPPayloadRegistry::RTPPayloadRegistry( RTPPayloadRegistry::~RTPPayloadRegistry() { while (!payload_type_map_.empty()) { - ModuleRTPUtility::PayloadTypeMap::iterator it = payload_type_map_.begin(); + RtpUtility::PayloadTypeMap::iterator it = payload_type_map_.begin(); delete it->second; payload_type_map_.erase(it); } @@ -69,12 +69,12 @@ int32_t RTPPayloadRegistry::RegisterReceivePayload( CriticalSectionScoped cs(crit_sect_.get()); - ModuleRTPUtility::PayloadTypeMap::iterator it = - payload_type_map_.find(payload_type); + RtpUtility::PayloadTypeMap::iterator it = + payload_type_map_.find(payload_type); if (it != payload_type_map_.end()) { // We already use this payload type. - ModuleRTPUtility::Payload* payload = it->second; + RtpUtility::Payload* payload = it->second; assert(payload); @@ -83,7 +83,7 @@ int32_t RTPPayloadRegistry::RegisterReceivePayload( // Check if it's the same as we already have. // If same, ignore sending an error. if (payload_name_length == name_length && - ModuleRTPUtility::StringCompare( + RtpUtility::StringCompare( payload->name, payload_name, payload_name_length)) { if (rtp_payload_strategy_->PayloadIsCompatible(*payload, frequency, channels, rate)) { @@ -100,18 +100,18 @@ int32_t RTPPayloadRegistry::RegisterReceivePayload( payload_name, payload_name_length, frequency, channels, rate); } - ModuleRTPUtility::Payload* payload = NULL; + RtpUtility::Payload* payload = NULL; // Save the RED payload type. Used in both audio and video. - if (ModuleRTPUtility::StringCompare(payload_name, "red", 3)) { + if (RtpUtility::StringCompare(payload_name, "red", 3)) { red_payload_type_ = payload_type; - payload = new ModuleRTPUtility::Payload; + payload = new RtpUtility::Payload; memset(payload, 0, sizeof(*payload)); payload->audio = false; strncpy(payload->name, payload_name, RTP_PAYLOAD_NAME_SIZE - 1); - } else if (ModuleRTPUtility::StringCompare(payload_name, "ulpfec", 3)) { + } else if (RtpUtility::StringCompare(payload_name, "ulpfec", 3)) { ulpfec_payload_type_ = payload_type; - payload = new ModuleRTPUtility::Payload; + payload = new RtpUtility::Payload; memset(payload, 0, sizeof(*payload)); payload->audio = false; strncpy(payload->name, payload_name, RTP_PAYLOAD_NAME_SIZE - 1); @@ -132,7 +132,7 @@ int32_t RTPPayloadRegistry::RegisterReceivePayload( int32_t RTPPayloadRegistry::DeRegisterReceivePayload( const int8_t payload_type) { CriticalSectionScoped cs(crit_sect_.get()); - ModuleRTPUtility::PayloadTypeMap::iterator it = + RtpUtility::PayloadTypeMap::iterator it = payload_type_map_.find(payload_type); assert(it != payload_type_map_.end()); delete it->second; @@ -149,15 +149,14 @@ void RTPPayloadRegistry::DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType( const uint32_t frequency, const uint8_t channels, const uint32_t rate) { - ModuleRTPUtility::PayloadTypeMap::iterator iterator = - payload_type_map_.begin(); + RtpUtility::PayloadTypeMap::iterator iterator = payload_type_map_.begin(); for (; iterator != payload_type_map_.end(); ++iterator) { - ModuleRTPUtility::Payload* payload = iterator->second; + RtpUtility::Payload* payload = iterator->second; size_t name_length = strlen(payload->name); - if (payload_name_length == name_length - && ModuleRTPUtility::StringCompare(payload->name, payload_name, - payload_name_length)) { + if (payload_name_length == name_length && + RtpUtility::StringCompare( + payload->name, payload_name, payload_name_length)) { // We found the payload name in the list. // If audio, check frequency and rate. if (payload->audio) { @@ -168,7 +167,7 @@ void RTPPayloadRegistry::DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType( payload_type_map_.erase(iterator); break; } - } else if (ModuleRTPUtility::StringCompare(payload_name, "red", 3)) { + } else if (RtpUtility::StringCompare(payload_name, "red", 3)) { delete payload; payload_type_map_.erase(iterator); break; @@ -188,16 +187,15 @@ int32_t RTPPayloadRegistry::ReceivePayloadType( CriticalSectionScoped cs(crit_sect_.get()); - ModuleRTPUtility::PayloadTypeMap::const_iterator it = - payload_type_map_.begin(); + RtpUtility::PayloadTypeMap::const_iterator it = payload_type_map_.begin(); for (; it != payload_type_map_.end(); ++it) { - ModuleRTPUtility::Payload* payload = it->second; + RtpUtility::Payload* payload = it->second; assert(payload); size_t name_length = strlen(payload->name); if (payload_name_length == name_length && - ModuleRTPUtility::StringCompare( + RtpUtility::StringCompare( payload->name, payload_name, payload_name_length)) { // Name matches. if (payload->audio) { @@ -261,9 +259,9 @@ bool RTPPayloadRegistry::RestoreOriginalPacket(uint8_t** restored_packet, *packet_length -= kRtxHeaderSize; // Replace the SSRC and the sequence number with the originals. - ModuleRTPUtility::AssignUWord16ToBuffer(*restored_packet + 2, - original_sequence_number); - ModuleRTPUtility::AssignUWord32ToBuffer(*restored_packet + 8, original_ssrc); + RtpUtility::AssignUWord16ToBuffer(*restored_packet + 2, + original_sequence_number); + RtpUtility::AssignUWord32ToBuffer(*restored_packet + 8, original_ssrc); CriticalSectionScoped cs(crit_sect_.get()); @@ -307,8 +305,8 @@ bool RTPPayloadRegistry::IsEncapsulated(const RTPHeader& header) const { bool RTPPayloadRegistry::GetPayloadSpecifics(uint8_t payload_type, PayloadUnion* payload) const { CriticalSectionScoped cs(crit_sect_.get()); - ModuleRTPUtility::PayloadTypeMap::const_iterator it = - payload_type_map_.find(payload_type); + RtpUtility::PayloadTypeMap::const_iterator it = + payload_type_map_.find(payload_type); // Check that this is a registered payload type. if (it == payload_type_map_.end()) { @@ -320,7 +318,7 @@ bool RTPPayloadRegistry::GetPayloadSpecifics(uint8_t payload_type, int RTPPayloadRegistry::GetPayloadTypeFrequency( uint8_t payload_type) const { - ModuleRTPUtility::Payload* payload; + RtpUtility::Payload* payload; if (!PayloadTypeToPayload(payload_type, payload)) { return -1; } @@ -329,12 +327,12 @@ int RTPPayloadRegistry::GetPayloadTypeFrequency( } bool RTPPayloadRegistry::PayloadTypeToPayload( - const uint8_t payload_type, - ModuleRTPUtility::Payload*& payload) const { + const uint8_t payload_type, + RtpUtility::Payload*& payload) const { CriticalSectionScoped cs(crit_sect_.get()); - ModuleRTPUtility::PayloadTypeMap::const_iterator it = - payload_type_map_.find(payload_type); + RtpUtility::PayloadTypeMap::const_iterator it = + payload_type_map_.find(payload_type); // Check that this is a registered payload type. if (it == payload_type_map_.end()) { @@ -365,11 +363,10 @@ class RTPPayloadAudioStrategy : public RTPPayloadStrategy { public: virtual bool CodecsMustBeUnique() const OVERRIDE { return true; } - virtual bool PayloadIsCompatible( - const ModuleRTPUtility::Payload& payload, - const uint32_t frequency, - const uint8_t channels, - const uint32_t rate) const OVERRIDE { + virtual bool PayloadIsCompatible(const RtpUtility::Payload& payload, + const uint32_t frequency, + const uint8_t channels, + const uint32_t rate) const OVERRIDE { return payload.audio && payload.typeSpecific.Audio.frequency == frequency && @@ -378,19 +375,18 @@ class RTPPayloadAudioStrategy : public RTPPayloadStrategy { payload.typeSpecific.Audio.rate == 0 || rate == 0); } - virtual void UpdatePayloadRate( - ModuleRTPUtility::Payload* payload, - const uint32_t rate) const OVERRIDE { + virtual void UpdatePayloadRate(RtpUtility::Payload* payload, + const uint32_t rate) const OVERRIDE { payload->typeSpecific.Audio.rate = rate; } - virtual ModuleRTPUtility::Payload* CreatePayloadType( + virtual RtpUtility::Payload* CreatePayloadType( const char payloadName[RTP_PAYLOAD_NAME_SIZE], const int8_t payloadType, const uint32_t frequency, const uint8_t channels, const uint32_t rate) const OVERRIDE { - ModuleRTPUtility::Payload* payload = new ModuleRTPUtility::Payload; + RtpUtility::Payload* payload = new RtpUtility::Payload; payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1); assert(frequency >= 1000); @@ -401,8 +397,7 @@ class RTPPayloadAudioStrategy : public RTPPayloadStrategy { return payload; } - int GetPayloadTypeFrequency( - const ModuleRTPUtility::Payload& payload) const { + int GetPayloadTypeFrequency(const RtpUtility::Payload& payload) const { return payload.typeSpecific.Audio.frequency; } }; @@ -411,37 +406,37 @@ class RTPPayloadVideoStrategy : public RTPPayloadStrategy { public: virtual bool CodecsMustBeUnique() const OVERRIDE { return false; } - virtual bool PayloadIsCompatible( - const ModuleRTPUtility::Payload& payload, - const uint32_t frequency, - const uint8_t channels, - const uint32_t rate) const OVERRIDE { + virtual bool PayloadIsCompatible(const RtpUtility::Payload& payload, + const uint32_t frequency, + const uint8_t channels, + const uint32_t rate) const OVERRIDE { return !payload.audio; } - virtual void UpdatePayloadRate( - ModuleRTPUtility::Payload* payload, - const uint32_t rate) const OVERRIDE { + virtual void UpdatePayloadRate(RtpUtility::Payload* payload, + const uint32_t rate) const OVERRIDE { payload->typeSpecific.Video.maxRate = rate; } - virtual ModuleRTPUtility::Payload* CreatePayloadType( + virtual RtpUtility::Payload* CreatePayloadType( const char payloadName[RTP_PAYLOAD_NAME_SIZE], const int8_t payloadType, const uint32_t frequency, const uint8_t channels, const uint32_t rate) const OVERRIDE { RtpVideoCodecTypes videoType = kRtpVideoGeneric; - if (ModuleRTPUtility::StringCompare(payloadName, "VP8", 3)) { + if (RtpUtility::StringCompare(payloadName, "VP8", 3)) { videoType = kRtpVideoVp8; - } else if (ModuleRTPUtility::StringCompare(payloadName, "I420", 4)) { + } else if (RtpUtility::StringCompare(payloadName, "H264", 4)) { + videoType = kRtpVideoH264; + } else if (RtpUtility::StringCompare(payloadName, "I420", 4)) { videoType = kRtpVideoGeneric; - } else if (ModuleRTPUtility::StringCompare(payloadName, "ULPFEC", 6)) { + } else if (RtpUtility::StringCompare(payloadName, "ULPFEC", 6)) { videoType = kRtpVideoNone; } else { videoType = kRtpVideoGeneric; } - ModuleRTPUtility::Payload* payload = new ModuleRTPUtility::Payload; + RtpUtility::Payload* payload = new RtpUtility::Payload; payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1); @@ -451,8 +446,7 @@ class RTPPayloadVideoStrategy : public RTPPayloadStrategy { return payload; } - int GetPayloadTypeFrequency( - const ModuleRTPUtility::Payload& payload) const { + int GetPayloadTypeFrequency(const RtpUtility::Payload& payload) const { return kVideoPayloadTypeFrequency; } }; diff --git a/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc b/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc index c03ffcd1..2dacbdd1 100644 --- a/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc @@ -36,18 +36,19 @@ class RtpPayloadRegistryTest : public ::testing::Test { } protected: - ModuleRTPUtility::Payload* ExpectReturnOfTypicalAudioPayload( - uint8_t payload_type, uint32_t rate) { + RtpUtility::Payload* ExpectReturnOfTypicalAudioPayload(uint8_t payload_type, + uint32_t rate) { bool audio = true; - ModuleRTPUtility::Payload returned_payload = { "name", audio, { - // Initialize the audio struct in this case. - { kTypicalFrequency, kTypicalChannels, rate } - }}; + RtpUtility::Payload returned_payload = { + "name", + audio, + {// Initialize the audio struct in this case. + {kTypicalFrequency, kTypicalChannels, rate}}}; // Note: we return a new payload since the payload registry takes ownership // of the created object. - ModuleRTPUtility::Payload* returned_payload_on_heap = - new ModuleRTPUtility::Payload(returned_payload); + RtpUtility::Payload* returned_payload_on_heap = + new RtpUtility::Payload(returned_payload); EXPECT_CALL(*mock_payload_strategy_, CreatePayloadType(kTypicalPayloadName, payload_type, kTypicalFrequency, @@ -62,7 +63,7 @@ class RtpPayloadRegistryTest : public ::testing::Test { TEST_F(RtpPayloadRegistryTest, RegistersAndRemembersPayloadsUntilDeregistered) { uint8_t payload_type = 97; - ModuleRTPUtility::Payload* returned_payload_on_heap = + RtpUtility::Payload* returned_payload_on_heap = ExpectReturnOfTypicalAudioPayload(payload_type, kTypicalRate); bool new_payload_created = false; @@ -72,7 +73,7 @@ TEST_F(RtpPayloadRegistryTest, RegistersAndRemembersPayloadsUntilDeregistered) { EXPECT_TRUE(new_payload_created) << "A new payload WAS created."; - ModuleRTPUtility::Payload* retrieved_payload = NULL; + RtpUtility::Payload* retrieved_payload = NULL; EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(payload_type, retrieved_payload)); @@ -99,7 +100,7 @@ TEST_F(RtpPayloadRegistryTest, DoesNotCreateNewPayloadTypeIfRed) { ASSERT_EQ(red_type_of_the_day, rtp_payload_registry_->red_payload_type()); - ModuleRTPUtility::Payload* retrieved_payload = NULL; + RtpUtility::Payload* retrieved_payload = NULL; EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(red_type_of_the_day, retrieved_payload)); EXPECT_FALSE(retrieved_payload->audio); @@ -111,7 +112,7 @@ TEST_F(RtpPayloadRegistryTest, uint8_t payload_type = 97; bool ignored = false; - ModuleRTPUtility::Payload* first_payload_on_heap = + RtpUtility::Payload* first_payload_on_heap = ExpectReturnOfTypicalAudioPayload(payload_type, kTypicalRate); EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload( kTypicalPayloadName, payload_type, kTypicalFrequency, kTypicalChannels, @@ -121,7 +122,7 @@ TEST_F(RtpPayloadRegistryTest, kTypicalPayloadName, payload_type, kTypicalFrequency, kTypicalChannels, kTypicalRate, &ignored)) << "Adding same codec twice = bad."; - ModuleRTPUtility::Payload* second_payload_on_heap = + RtpUtility::Payload* second_payload_on_heap = ExpectReturnOfTypicalAudioPayload(payload_type - 1, kTypicalRate); EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload( kTypicalPayloadName, payload_type - 1, kTypicalFrequency, @@ -129,7 +130,7 @@ TEST_F(RtpPayloadRegistryTest, "With a different payload type is fine though."; // Ensure both payloads are preserved. - ModuleRTPUtility::Payload* retrieved_payload = NULL; + RtpUtility::Payload* retrieved_payload = NULL; EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(payload_type, retrieved_payload)); EXPECT_EQ(first_payload_on_heap, retrieved_payload); @@ -168,7 +169,7 @@ TEST_F(RtpPayloadRegistryTest, kTypicalPayloadName, payload_type - 1, kTypicalFrequency, kTypicalChannels, kTypicalRate, &ignored)); - ModuleRTPUtility::Payload* retrieved_payload = NULL; + RtpUtility::Payload* retrieved_payload = NULL; EXPECT_FALSE(rtp_payload_registry_->PayloadTypeToPayload( payload_type, retrieved_payload)) << "The first payload should be " "deregistered because the only thing that differs is payload type."; diff --git a/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/modules/rtp_rtcp/source/rtp_receiver_audio.cc index c8104cc3..05eefbe0 100644 --- a/modules/rtp_rtcp/source/rtp_receiver_audio.cc +++ b/modules/rtp_rtcp/source/rtp_receiver_audio.cc @@ -159,10 +159,10 @@ int32_t RTPReceiverAudio::OnNewPayloadTypeCreated( uint32_t frequency) { CriticalSectionScoped lock(crit_sect_.get()); - if (ModuleRTPUtility::StringCompare(payload_name, "telephone-event", 15)) { + if (RtpUtility::StringCompare(payload_name, "telephone-event", 15)) { telephone_event_payload_type_ = payload_type; } - if (ModuleRTPUtility::StringCompare(payload_name, "cn", 2)) { + if (RtpUtility::StringCompare(payload_name, "cn", 2)) { // we can have three CNG on 8000Hz, 16000Hz and 32000Hz if (frequency == 8000) { cng_nb_payload_type_ = payload_type; diff --git a/modules/rtp_rtcp/source/rtp_receiver_audio.h b/modules/rtp_rtcp/source/rtp_receiver_audio.h index 0ffd4bf4..4fb7256d 100644 --- a/modules/rtp_rtcp/source/rtp_receiver_audio.h +++ b/modules/rtp_rtcp/source/rtp_receiver_audio.h @@ -83,7 +83,7 @@ class RTPReceiverAudio : public RTPReceiverStrategy, // We do not allow codecs to have multiple payload types for audio, so we // need to override the default behavior (which is to do nothing). void PossiblyRemoveExistingPayloadType( - ModuleRTPUtility::PayloadTypeMap* payload_type_map, + RtpUtility::PayloadTypeMap* payload_type_map, const char payload_name[RTP_PAYLOAD_NAME_SIZE], size_t payload_name_length, uint32_t frequency, diff --git a/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/modules/rtp_rtcp/source/rtp_receiver_impl.cc index d92618f2..7493488d 100644 --- a/modules/rtp_rtcp/source/rtp_receiver_impl.cc +++ b/modules/rtp_rtcp/source/rtp_receiver_impl.cc @@ -22,10 +22,10 @@ namespace webrtc { -using ModuleRTPUtility::GetCurrentRTP; -using ModuleRTPUtility::Payload; -using ModuleRTPUtility::RTPPayloadParser; -using ModuleRTPUtility::StringCompare; +using RtpUtility::GetCurrentRTP; +using RtpUtility::Payload; +using RtpUtility::RTPPayloadParser; +using RtpUtility::StringCompare; RtpReceiver* RtpReceiver::CreateVideoReceiver( int id, Clock* clock, diff --git a/modules/rtp_rtcp/source/rtp_receiver_video.cc b/modules/rtp_rtcp/source/rtp_receiver_video.cc index 5bb519f6..c058ed5d 100644 --- a/modules/rtp_rtcp/source/rtp_receiver_video.cc +++ b/modules/rtp_rtcp/source/rtp_receiver_video.cc @@ -113,6 +113,8 @@ int32_t RTPReceiverVideo::ParseVideoCodecSpecific( return ReceiveGenericCodec(rtp_header, payload_data, payload_data_length); case kRtpVideoVp8: return ReceiveVp8Codec(rtp_header, payload_data, payload_data_length); + case kRtpVideoH264: + assert(false); // Not yet supported. case kRtpVideoNone: break; } @@ -127,12 +129,11 @@ int32_t RTPReceiverVideo::BuildRTPheader( if (rtp_header->header.markerBit) { data_buffer[1] |= kRtpMarkerBitMask; // MarkerBit is 1 } - ModuleRTPUtility::AssignUWord16ToBuffer(data_buffer + 2, - rtp_header->header.sequenceNumber); - ModuleRTPUtility::AssignUWord32ToBuffer(data_buffer + 4, - rtp_header->header.timestamp); - ModuleRTPUtility::AssignUWord32ToBuffer(data_buffer + 8, - rtp_header->header.ssrc); + RtpUtility::AssignUWord16ToBuffer(data_buffer + 2, + rtp_header->header.sequenceNumber); + RtpUtility::AssignUWord32ToBuffer(data_buffer + 4, + rtp_header->header.timestamp); + RtpUtility::AssignUWord32ToBuffer(data_buffer + 8, rtp_header->header.ssrc); int32_t rtp_header_length = 12; @@ -144,8 +145,7 @@ int32_t RTPReceiverVideo::BuildRTPheader( } uint8_t* ptr = &data_buffer[rtp_header_length]; for (uint32_t i = 0; i < rtp_header->header.numCSRCs; ++i) { - ModuleRTPUtility::AssignUWord32ToBuffer(ptr, - rtp_header->header.arrOfCSRCs[i]); + RtpUtility::AssignUWord32ToBuffer(ptr, rtp_header->header.arrOfCSRCs[i]); ptr += 4; } data_buffer[0] = (data_buffer[0] & 0xf0) | rtp_header->header.numCSRCs; @@ -158,8 +158,8 @@ int32_t RTPReceiverVideo::BuildRTPheader( int32_t RTPReceiverVideo::ReceiveVp8Codec(WebRtcRTPHeader* rtp_header, const uint8_t* payload_data, uint16_t payload_data_length) { - ModuleRTPUtility::RTPPayload parsed_packet; - ModuleRTPUtility::RTPPayloadParser rtp_payload_parser( + RtpUtility::RTPPayload parsed_packet; + RtpUtility::RTPPayloadParser rtp_payload_parser( kRtpVideoVp8, payload_data, payload_data_length); if (!rtp_payload_parser.Parse(parsed_packet)) @@ -168,11 +168,12 @@ int32_t RTPReceiverVideo::ReceiveVp8Codec(WebRtcRTPHeader* rtp_header, if (parsed_packet.info.VP8.dataLength == 0) return 0; - rtp_header->frameType = (parsed_packet.frameType == ModuleRTPUtility::kIFrame) - ? kVideoFrameKey : kVideoFrameDelta; + rtp_header->frameType = (parsed_packet.frameType == RtpUtility::kIFrame) + ? kVideoFrameKey + : kVideoFrameDelta; RTPVideoHeaderVP8* to_header = &rtp_header->type.Video.codecHeader.VP8; - ModuleRTPUtility::RTPPayloadVP8* from_header = &parsed_packet.info.VP8; + RtpUtility::RTPPayloadVP8* from_header = &parsed_packet.info.VP8; rtp_header->type.Video.isFirstPacket = from_header->beginningOfPartition && (from_header->partitionID == 0); diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc index 70fe7174..855d51b7 100644 --- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc +++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc @@ -37,7 +37,8 @@ RtpRtcp::Configuration::Configuration() rtt_stats(NULL), audio_messages(NullObjectRtpAudioFeedback()), remote_bitrate_estimator(NULL), - paced_sender(NULL) { + paced_sender(NULL), + send_bitrate_observer(NULL) { } RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) { @@ -60,7 +61,8 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration) configuration.clock, configuration.outgoing_transport, configuration.audio_messages, - configuration.paced_sender), + configuration.paced_sender, + configuration.send_bitrate_observer), rtcp_sender_(configuration.id, configuration.audio, configuration.clock, @@ -333,6 +335,42 @@ int32_t ModuleRtpRtcpImpl::SetSequenceNumber( return 0; // TODO(pwestin): change to void. } +void ModuleRtpRtcpImpl::SetRtpStateForSsrc(uint32_t ssrc, + const RtpState& rtp_state) { + if (rtp_sender_.SSRC() == ssrc) { + rtp_sender_.SetRtpState(rtp_state); + return; + } + if (rtp_sender_.RtxSsrc() == ssrc) { + rtp_sender_.SetRtxRtpState(rtp_state); + return; + } + + CriticalSectionScoped lock(critical_section_module_ptrs_.get()); + for (size_t i = 0; i < child_modules_.size(); ++i) { + child_modules_[i]->SetRtpStateForSsrc(ssrc, rtp_state); + } +} + +bool ModuleRtpRtcpImpl::GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) { + if (rtp_sender_.SSRC() == ssrc) { + *rtp_state = rtp_sender_.GetRtpState(); + return true; + } + + if (rtp_sender_.RtxSsrc() == ssrc) { + *rtp_state = rtp_sender_.GetRtxRtpState(); + return true; + } + + CriticalSectionScoped lock(critical_section_module_ptrs_.get()); + for (size_t i = 0; i < child_modules_.size(); ++i) { + if (child_modules_[i]->GetRtpStateForSsrc(ssrc, rtp_state)) + return true; + } + return false; +} + uint32_t ModuleRtpRtcpImpl::SSRC() const { return rtp_sender_.SSRC(); } @@ -1198,16 +1236,6 @@ void ModuleRtpRtcpImpl::BitrateSent(uint32_t* total_rate, *nack_rate = rtp_sender_.NackOverheadRate(); } -void ModuleRtpRtcpImpl::RegisterVideoBitrateObserver( - BitrateStatisticsObserver* observer) { - assert(!IsDefaultModule()); - rtp_sender_.RegisterBitrateObserver(observer); -} - -BitrateStatisticsObserver* ModuleRtpRtcpImpl::GetVideoBitrateObserver() const { - return rtp_sender_.GetBitrateObserver(); -} - void ModuleRtpRtcpImpl::OnRequestIntraFrame() { RequestKeyFrame(); } diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/modules/rtp_rtcp/source/rtp_rtcp_impl.h index 55826b6f..b65131fb 100644 --- a/modules/rtp_rtcp/source/rtp_rtcp_impl.h +++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.h @@ -73,6 +73,10 @@ class ModuleRtpRtcpImpl : public RtpRtcp { // Set SequenceNumber, default is a random number. virtual int32_t SetSequenceNumber(const uint16_t seq) OVERRIDE; + virtual void SetRtpStateForSsrc(uint32_t ssrc, + const RtpState& rtp_state) OVERRIDE; + virtual bool GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) OVERRIDE; + virtual uint32_t SSRC() const OVERRIDE; // Configure SSRC, default is a random number. @@ -340,11 +344,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp { uint32_t* fec_rate, uint32_t* nackRate) const OVERRIDE; - virtual void RegisterVideoBitrateObserver(BitrateStatisticsObserver* observer) - OVERRIDE; - - virtual BitrateStatisticsObserver* GetVideoBitrateObserver() const OVERRIDE; - virtual uint32_t SendTimeOfSendReport(const uint32_t send_report); virtual bool SendTimeOfXrRrReport(uint32_t mid_ntp, int64_t* time_ms) const; diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc index eb76cfe7..930778c3 100644 --- a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc @@ -232,8 +232,9 @@ class RtpSendingTestTransport : public Transport { virtual int SendPacket(int channel, const void* data, int length) { RTPHeader header; scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create()); - EXPECT_TRUE( - parser->Parse(static_cast<const uint8_t*>(data), length, &header)); + EXPECT_TRUE(parser->Parse(static_cast<const uint8_t*>(data), + static_cast<size_t>(length), + &header)); bytes_received_[header.ssrc] += length; ++packets_received_[header.ssrc]; return length; diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc index 74947693..858fc42a 100644 --- a/modules/rtp_rtcp/source/rtp_sender.cc +++ b/modules/rtp_rtcp/source/rtp_sender.cc @@ -16,6 +16,7 @@ #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/logging.h" +#include "webrtc/system_wrappers/interface/tick_util.h" #include "webrtc/system_wrappers/interface/trace_event.h" namespace webrtc { @@ -44,7 +45,8 @@ RTPSender::RTPSender(const int32_t id, Clock* clock, Transport* transport, RtpAudioFeedback* audio_feedback, - PacedSender* paced_sender) + PacedSender* paced_sender, + BitrateStatisticsObserver* bitrate_callback) : clock_(clock), bitrate_sent_(clock, this), id_(id), @@ -71,10 +73,10 @@ RTPSender::RTPSender(const int32_t id, statistics_crit_(CriticalSectionWrapper::CreateCriticalSection()), frame_count_observer_(NULL), rtp_stats_callback_(NULL), - bitrate_callback_(NULL), + bitrate_callback_(bitrate_callback), // RTP variables - start_time_stamp_forced_(false), - start_time_stamp_(0), + start_timestamp_forced_(false), + start_timestamp_(0), ssrc_db_(*SSRCDatabase::GetSSRCDatabase()), remote_ssrc_(0), sequence_number_forced_(false), @@ -118,7 +120,7 @@ RTPSender::~RTPSender() { SSRCDatabase::ReturnSSRCDatabase(); delete send_critsect_; while (!payload_type_map_.empty()) { - std::map<int8_t, ModuleRTPUtility::Payload *>::iterator it = + std::map<int8_t, RtpUtility::Payload*>::iterator it = payload_type_map_.begin(); delete it->second; payload_type_map_.erase(it); @@ -223,17 +225,17 @@ int32_t RTPSender::RegisterPayload( assert(payload_name); CriticalSectionScoped cs(send_critsect_); - std::map<int8_t, ModuleRTPUtility::Payload *>::iterator it = + std::map<int8_t, RtpUtility::Payload*>::iterator it = payload_type_map_.find(payload_number); if (payload_type_map_.end() != it) { // We already use this payload type. - ModuleRTPUtility::Payload *payload = it->second; + RtpUtility::Payload* payload = it->second; assert(payload); // Check if it's the same as we already have. - if (ModuleRTPUtility::StringCompare(payload->name, payload_name, - RTP_PAYLOAD_NAME_SIZE - 1)) { + if (RtpUtility::StringCompare( + payload->name, payload_name, RTP_PAYLOAD_NAME_SIZE - 1)) { if (audio_configured_ && payload->audio && payload->typeSpecific.Audio.frequency == frequency && (payload->typeSpecific.Audio.rate == rate || @@ -249,7 +251,7 @@ int32_t RTPSender::RegisterPayload( return -1; } int32_t ret_val = -1; - ModuleRTPUtility::Payload *payload = NULL; + RtpUtility::Payload* payload = NULL; if (audio_configured_) { ret_val = audio_->RegisterAudioPayload(payload_name, payload_number, frequency, channels, rate, payload); @@ -267,13 +269,13 @@ int32_t RTPSender::DeRegisterSendPayload( const int8_t payload_type) { CriticalSectionScoped lock(send_critsect_); - std::map<int8_t, ModuleRTPUtility::Payload *>::iterator it = + std::map<int8_t, RtpUtility::Payload*>::iterator it = payload_type_map_.find(payload_type); if (payload_type_map_.end() == it) { return -1; } - ModuleRTPUtility::Payload *payload = it->second; + RtpUtility::Payload* payload = it->second; delete payload; payload_type_map_.erase(it); return 0; @@ -303,12 +305,17 @@ int32_t RTPSender::SetMaxPayloadLength( } uint16_t RTPSender::MaxDataPayloadLength() const { + int rtx; + { + CriticalSectionScoped rtx_lock(send_critsect_); + rtx = rtx_; + } if (audio_configured_) { return max_payload_length_ - RTPHeaderLength(); } else { return max_payload_length_ - RTPHeaderLength() // RTP overhead. - video_->FECPacketOverhead() // FEC/ULP/RED overhead. - - ((rtx_) ? 2 : 0); // RTX overhead. + - ((rtx) ? 2 : 0); // RTX overhead. } } @@ -328,6 +335,11 @@ void RTPSender::SetRtxSsrc(uint32_t ssrc) { ssrc_rtx_ = ssrc; } +uint32_t RTPSender::RtxSsrc() const { + CriticalSectionScoped cs(send_critsect_); + return ssrc_rtx_; +} + void RTPSender::RTXStatus(int* mode, uint32_t* ssrc, int* payload_type) const { CriticalSectionScoped cs(send_critsect_); @@ -365,14 +377,14 @@ int32_t RTPSender::CheckPayloadType(const int8_t payload_type, } return 0; } - std::map<int8_t, ModuleRTPUtility::Payload *>::iterator it = + std::map<int8_t, RtpUtility::Payload*>::iterator it = payload_type_map_.find(payload_type); if (it == payload_type_map_.end()) { LOG(LS_WARNING) << "Payload type " << payload_type << " not registered."; return -1; } payload_type_ = payload_type; - ModuleRTPUtility::Payload *payload = it->second; + RtpUtility::Payload* payload = it->second; assert(payload); if (!payload->audio && !audio_configured_) { video_->SetVideoCodecType(payload->typeSpecific.Video.videoCodecType); @@ -388,9 +400,11 @@ int32_t RTPSender::SendOutgoingData( const uint8_t *payload_data, const uint32_t payload_size, const RTPFragmentationHeader *fragmentation, VideoCodecInformation *codec_info, const RTPVideoTypeHeader *rtp_type_hdr) { + uint32_t ssrc; { // Drop this packet if we're not sending media packets. CriticalSectionScoped cs(send_critsect_); + ssrc = ssrc_; if (!sending_media_) { return 0; } @@ -434,17 +448,13 @@ int32_t RTPSender::SendOutgoingData( CriticalSectionScoped cs(statistics_crit_.get()); uint32_t frame_count = ++frame_counts_[frame_type]; if (frame_count_observer_) { - frame_count_observer_->FrameCountUpdated(frame_type, - frame_count, - ssrc_); + frame_count_observer_->FrameCountUpdated(frame_type, frame_count, ssrc); } return ret_val; } int RTPSender::SendRedundantPayloads(int payload_type, int bytes_to_send) { - if (!(rtx_ & kRtxRedundantPayloads)) - return 0; uint8_t buffer[IP_PACKET_SIZE]; int bytes_left = bytes_to_send; while (bytes_left > 0) { @@ -456,7 +466,7 @@ int RTPSender::SendRedundantPayloads(int payload_type, int bytes_to_send) { } if (!PrepareAndSendPacket(buffer, length, capture_time_ms, true, false)) return -1; - ModuleRTPUtility::RTPHeaderParser rtp_parser(buffer, length); + RtpUtility::RtpHeaderParser rtp_parser(buffer, length); RTPHeader rtp_header; rtp_parser.Parse(rtp_header); bytes_left -= length - rtp_header.headerLength; @@ -492,7 +502,7 @@ bool RTPSender::SendPaddingAccordingToBitrate( CriticalSectionScoped cs(send_critsect_); // Add the random RTP timestamp offset and store the capture time for // later calculation of the send time offset. - timestamp = start_time_stamp_ + capture_timestamp; + timestamp = start_timestamp_ + capture_timestamp; timestamp_ = timestamp; capture_time_ms_ = capture_time_ms; last_timestamp_time_ms_ = clock_->TimeInMilliseconds(); @@ -566,6 +576,7 @@ int RTPSender::SendPadData(int payload_type, uint32_t timestamp, ++sequence_number_rtx_; } } + uint8_t padding_packet[IP_PACKET_SIZE]; int header_length = CreateRTPHeader(padding_packet, payload_type, ssrc, false, timestamp, sequence_number, NULL, @@ -604,16 +615,21 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, uint32_t min_resend_time) { } if (paced_sender_) { - ModuleRTPUtility::RTPHeaderParser rtp_parser(data_buffer, length); + RtpUtility::RtpHeaderParser rtp_parser(data_buffer, length); RTPHeader header; if (!rtp_parser.Parse(header)) { assert(false); return -1; } + // Convert from TickTime to Clock since capture_time_ms is based on + // TickTime. + // TODO(holmer): Remove this conversion when we remove the use of TickTime. + int64_t clock_delta_ms = clock_->TimeInMilliseconds() - + TickTime::MillisecondTimestamp(); if (!paced_sender_->SendPacket(PacedSender::kHighPriority, header.ssrc, header.sequenceNumber, - capture_time_ms, + capture_time_ms + clock_delta_ms, length - header.headerLength, true)) { // We can't send the packet right now. @@ -622,6 +638,7 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, uint32_t min_resend_time) { } } + CriticalSectionScoped lock(send_critsect_); return PrepareAndSendPacket(data_buffer, length, capture_time_ms, (rtx_ & kRtxRetransmitted) > 0, true) ? length : -1; @@ -778,8 +795,15 @@ bool RTPSender::TimeToSendPacket(uint16_t sequence_number, if (!retransmission && capture_time_ms > 0) { UpdateDelayStatistics(capture_time_ms, clock_->TimeInMilliseconds()); } - return PrepareAndSendPacket(data_buffer, length, capture_time_ms, - retransmission && (rtx_ & kRtxRetransmitted) > 0, + int rtx; + { + CriticalSectionScoped lock(send_critsect_); + rtx = rtx_; + } + return PrepareAndSendPacket(data_buffer, + length, + capture_time_ms, + retransmission && (rtx & kRtxRetransmitted) > 0, retransmission); } @@ -790,7 +814,7 @@ bool RTPSender::PrepareAndSendPacket(uint8_t* buffer, bool is_retransmit) { uint8_t *buffer_to_send_ptr = buffer; - ModuleRTPUtility::RTPHeaderParser rtp_parser(buffer, length); + RtpUtility::RtpHeaderParser rtp_parser(buffer, length); RTPHeader rtp_header; rtp_parser.Parse(rtp_header); TRACE_EVENT_INSTANT2("webrtc_rtp", "PrepareAndSendPacket", @@ -821,12 +845,11 @@ void RTPSender::UpdateRtpStats(const uint8_t* buffer, bool is_retransmit) { StreamDataCounters* counters; // Get ssrc before taking statistics_crit_ to avoid possible deadlock. - uint32_t ssrc = SSRC(); + uint32_t ssrc = is_rtx ? RtxSsrc() : SSRC(); CriticalSectionScoped lock(statistics_crit_.get()); if (is_rtx) { counters = &rtx_rtp_stats_; - ssrc = ssrc_rtx_; } else { counters = &rtp_stats_; } @@ -868,6 +891,7 @@ int RTPSender::TimeToSendPadding(int bytes) { int payload_type; int64_t capture_time_ms; uint32_t timestamp; + int rtx; { CriticalSectionScoped cs(send_critsect_); if (!sending_media_) { @@ -883,8 +907,11 @@ int RTPSender::TimeToSendPadding(int bytes) { capture_time_ms += (clock_->TimeInMilliseconds() - last_timestamp_time_ms_); } + rtx = rtx_; } - int bytes_sent = SendRedundantPayloads(payload_type, bytes); + int bytes_sent = 0; + if ((rtx & kRtxRedundantPayloads) != 0) + bytes_sent = SendRedundantPayloads(payload_type, bytes); bytes -= bytes_sent; if (bytes > 0) { int padding_sent = SendPadData(payload_type, @@ -893,19 +920,19 @@ int RTPSender::TimeToSendPadding(int bytes) { bytes, kDontStore, true, - rtx_ == kRtxOff); + rtx == kRtxOff); bytes_sent += padding_sent; } return bytes_sent; } -// TODO(pwestin): send in the RTPHeaderParser to avoid parsing it again. +// TODO(pwestin): send in the RtpHeaderParser to avoid parsing it again. int32_t RTPSender::SendToNetwork( uint8_t *buffer, int payload_length, int rtp_header_length, int64_t capture_time_ms, StorageType storage, PacedSender::Priority priority) { - ModuleRTPUtility::RTPHeaderParser rtp_parser( - buffer, payload_length + rtp_header_length); + RtpUtility::RtpHeaderParser rtp_parser(buffer, + payload_length + rtp_header_length); RTPHeader rtp_header; rtp_parser.Parse(rtp_header); @@ -930,8 +957,11 @@ int32_t RTPSender::SendToNetwork( } if (paced_sender_ && storage != kDontStore) { + int64_t clock_delta_ms = clock_->TimeInMilliseconds() - + TickTime::MillisecondTimestamp(); if (!paced_sender_->SendPacket(priority, rtp_header.ssrc, - rtp_header.sequenceNumber, capture_time_ms, + rtp_header.sequenceNumber, + capture_time_ms + clock_delta_ms, payload_length, false)) { // We can't send the packet right now. // We will be called when it is time. @@ -966,6 +996,7 @@ void RTPSender::ProcessBitrate() { } uint16_t RTPSender::RTPHeaderLength() const { + CriticalSectionScoped lock(send_critsect_); uint16_t rtp_header_length = 12; if (include_csrcs_) { rtp_header_length += sizeof(uint32_t) * num_csrcs_; @@ -980,12 +1011,19 @@ uint16_t RTPSender::IncrementSequenceNumber() { } void RTPSender::ResetDataCounters() { + uint32_t ssrc; + uint32_t ssrc_rtx; + { + CriticalSectionScoped ssrc_lock(send_critsect_); + ssrc = ssrc_; + ssrc_rtx = ssrc_rtx_; + } CriticalSectionScoped lock(statistics_crit_.get()); rtp_stats_ = StreamDataCounters(); rtx_rtp_stats_ = StreamDataCounters(); if (rtp_stats_callback_) { - rtp_stats_callback_->DataCountersUpdated(rtp_stats_, ssrc_); - rtp_stats_callback_->DataCountersUpdated(rtx_rtp_stats_, ssrc_rtx_); + rtp_stats_callback_->DataCountersUpdated(rtp_stats_, ssrc); + rtp_stats_callback_->DataCountersUpdated(rtx_rtp_stats_, ssrc_rtx); } } @@ -1009,9 +1047,9 @@ int RTPSender::CreateRTPHeader( if (marker_bit) { header[1] |= kRtpMarkerBitMask; // Marker bit is set. } - ModuleRTPUtility::AssignUWord16ToBuffer(header + 2, sequence_number); - ModuleRTPUtility::AssignUWord32ToBuffer(header + 4, timestamp); - ModuleRTPUtility::AssignUWord32ToBuffer(header + 8, ssrc); + RtpUtility::AssignUWord16ToBuffer(header + 2, sequence_number); + RtpUtility::AssignUWord32ToBuffer(header + 4, timestamp); + RtpUtility::AssignUWord32ToBuffer(header + 8, ssrc); int32_t rtp_header_length = 12; // Add the CSRCs if any. @@ -1023,7 +1061,7 @@ int RTPSender::CreateRTPHeader( } uint8_t *ptr = &header[rtp_header_length]; for (int i = 0; i < num_csrcs; ++i) { - ModuleRTPUtility::AssignUWord32ToBuffer(ptr, csrcs[i]); + RtpUtility::AssignUWord32ToBuffer(ptr, csrcs[i]); ptr += 4; } header[0] = (header[0] & 0xf0) | num_csrcs; @@ -1040,16 +1078,18 @@ int RTPSender::CreateRTPHeader( return rtp_header_length; } -int32_t RTPSender::BuildRTPheader( - uint8_t *data_buffer, const int8_t payload_type, - const bool marker_bit, const uint32_t capture_timestamp, - int64_t capture_time_ms, const bool time_stamp_provided, - const bool inc_sequence_number) { +int32_t RTPSender::BuildRTPheader(uint8_t* data_buffer, + const int8_t payload_type, + const bool marker_bit, + const uint32_t capture_timestamp, + int64_t capture_time_ms, + const bool timestamp_provided, + const bool inc_sequence_number) { assert(payload_type >= 0); CriticalSectionScoped cs(send_critsect_); - if (time_stamp_provided) { - timestamp_ = start_time_stamp_ + capture_timestamp; + if (timestamp_provided) { + timestamp_ = start_timestamp_ + capture_timestamp; } else { // Make a unique time stamp. // We can't inc by the actual time, since then we increase the risk of back @@ -1084,8 +1124,7 @@ uint16_t RTPSender::BuildRTPHeaderExtension(uint8_t* data_buffer) const { const uint32_t kHeaderLength = kRtpOneByteHeaderLength; // Add extension ID (0xBEDE). - ModuleRTPUtility::AssignUWord16ToBuffer(data_buffer, - kRtpOneByteHeaderExtensionId); + RtpUtility::AssignUWord16ToBuffer(data_buffer, kRtpOneByteHeaderExtensionId); // Add extensions. uint16_t total_block_length = 0; @@ -1118,8 +1157,8 @@ uint16_t RTPSender::BuildRTPHeaderExtension(uint8_t* data_buffer) const { } // Set header length (in number of Word32, header excluded). assert(total_block_length % 4 == 0); - ModuleRTPUtility::AssignUWord16ToBuffer(data_buffer + kPosLength, - total_block_length / 4); + RtpUtility::AssignUWord16ToBuffer(data_buffer + kPosLength, + total_block_length / 4); // Total added length. return kHeaderLength + total_block_length; } @@ -1153,8 +1192,8 @@ uint8_t RTPSender::BuildTransmissionTimeOffsetExtension( size_t pos = 0; const uint8_t len = 2; data_buffer[pos++] = (id << 4) + len; - ModuleRTPUtility::AssignUWord24ToBuffer(data_buffer + pos, - transmission_time_offset_); + RtpUtility::AssignUWord24ToBuffer(data_buffer + pos, + transmission_time_offset_); pos += 3; assert(pos == kTransmissionTimeOffsetLength); return kTransmissionTimeOffsetLength; @@ -1222,8 +1261,7 @@ uint8_t RTPSender::BuildAbsoluteSendTimeExtension(uint8_t* data_buffer) const { size_t pos = 0; const uint8_t len = 2; data_buffer[pos++] = (id << 4) + len; - ModuleRTPUtility::AssignUWord24ToBuffer(data_buffer + pos, - absolute_send_time_); + RtpUtility::AssignUWord24ToBuffer(data_buffer + pos, absolute_send_time_); pos += 3; assert(pos == kAbsoluteSendTimeLength); return kAbsoluteSendTimeLength; @@ -1271,8 +1309,8 @@ void RTPSender::UpdateTransmissionTimeOffset( return; } // Update transmission offset field (converting to a 90 kHz timestamp). - ModuleRTPUtility::AssignUWord24ToBuffer(rtp_packet + block_pos + 1, - time_diff_ms * 90); // RTP timestamp. + RtpUtility::AssignUWord24ToBuffer(rtp_packet + block_pos + 1, + time_diff_ms * 90); // RTP timestamp. } bool RTPSender::UpdateAudioLevel(uint8_t *rtp_packet, @@ -1359,18 +1397,19 @@ void RTPSender::UpdateAbsoluteSendTime( } // Update absolute send time field (convert ms to 24-bit unsigned with 18 bit // fractional part). - ModuleRTPUtility::AssignUWord24ToBuffer(rtp_packet + block_pos + 1, - ((now_ms << 18) / 1000) & 0x00ffffff); + RtpUtility::AssignUWord24ToBuffer(rtp_packet + block_pos + 1, + ((now_ms << 18) / 1000) & 0x00ffffff); } void RTPSender::SetSendingStatus(bool enabled) { if (enabled) { uint32_t frequency_hz = SendPayloadFrequency(); - uint32_t RTPtime = ModuleRTPUtility::GetCurrentRTP(clock_, frequency_hz); + uint32_t RTPtime = RtpUtility::GetCurrentRTP(clock_, frequency_hz); // Will be ignored if it's already configured via API. SetStartTimestamp(RTPtime, false); } else { + CriticalSectionScoped lock(send_critsect_); if (!ssrc_forced_) { // Generate a new SSRC. ssrc_db_.ReturnSSRC(ssrc_); @@ -1403,18 +1442,18 @@ uint32_t RTPSender::Timestamp() const { void RTPSender::SetStartTimestamp(uint32_t timestamp, bool force) { CriticalSectionScoped cs(send_critsect_); if (force) { - start_time_stamp_forced_ = force; - start_time_stamp_ = timestamp; + start_timestamp_forced_ = true; + start_timestamp_ = timestamp; } else { - if (!start_time_stamp_forced_) { - start_time_stamp_ = timestamp; + if (!start_timestamp_forced_) { + start_timestamp_ = timestamp; } } } uint32_t RTPSender::StartTimestamp() const { CriticalSectionScoped cs(send_critsect_); - return start_time_stamp_; + return start_timestamp_; } uint32_t RTPSender::GenerateNewSSRC() { @@ -1451,6 +1490,7 @@ uint32_t RTPSender::SSRC() const { } void RTPSender::SetCSRCStatus(const bool include) { + CriticalSectionScoped lock(send_critsect_); include_csrcs_ = include; } @@ -1589,8 +1629,8 @@ void RTPSender::BuildRtxPacket(uint8_t* buffer, uint16_t* length, CriticalSectionScoped cs(send_critsect_); uint8_t* data_buffer_rtx = buffer_rtx; // Add RTX header. - ModuleRTPUtility::RTPHeaderParser rtp_parser( - reinterpret_cast<const uint8_t *>(buffer), *length); + RtpUtility::RtpHeaderParser rtp_parser( + reinterpret_cast<const uint8_t*>(buffer), *length); RTPHeader rtp_header; rtp_parser.Parse(rtp_header); @@ -1607,15 +1647,15 @@ void RTPSender::BuildRtxPacket(uint8_t* buffer, uint16_t* length, // Replace sequence number. uint8_t *ptr = data_buffer_rtx + 2; - ModuleRTPUtility::AssignUWord16ToBuffer(ptr, sequence_number_rtx_++); + RtpUtility::AssignUWord16ToBuffer(ptr, sequence_number_rtx_++); // Replace SSRC. ptr += 6; - ModuleRTPUtility::AssignUWord32ToBuffer(ptr, ssrc_rtx_); + RtpUtility::AssignUWord32ToBuffer(ptr, ssrc_rtx_); // Add OSN (original sequence number). ptr = data_buffer_rtx + rtp_header.headerLength; - ModuleRTPUtility::AssignUWord16ToBuffer(ptr, rtp_header.sequenceNumber); + RtpUtility::AssignUWord16ToBuffer(ptr, rtp_header.sequenceNumber); ptr += 2; // Add original payload data. @@ -1626,8 +1666,6 @@ void RTPSender::BuildRtxPacket(uint8_t* buffer, uint16_t* length, void RTPSender::RegisterFrameCountObserver(FrameCountObserver* observer) { CriticalSectionScoped cs(statistics_crit_.get()); - if (observer != NULL) - assert(frame_count_observer_ == NULL); frame_count_observer_ = observer; } @@ -1639,8 +1677,6 @@ FrameCountObserver* RTPSender::GetFrameCountObserver() const { void RTPSender::RegisterRtpStatisticsCallback( StreamDataCountersCallback* callback) { CriticalSectionScoped cs(statistics_crit_.get()); - if (callback != NULL) - assert(rtp_stats_callback_ == NULL); rtp_stats_callback_ = callback; } @@ -1649,24 +1685,55 @@ StreamDataCountersCallback* RTPSender::GetRtpStatisticsCallback() const { return rtp_stats_callback_; } -void RTPSender::RegisterBitrateObserver(BitrateStatisticsObserver* observer) { - CriticalSectionScoped cs(statistics_crit_.get()); - if (observer != NULL) - assert(bitrate_callback_ == NULL); - bitrate_callback_ = observer; -} - -BitrateStatisticsObserver* RTPSender::GetBitrateObserver() const { - CriticalSectionScoped cs(statistics_crit_.get()); - return bitrate_callback_; -} - uint32_t RTPSender::BitrateSent() const { return bitrate_sent_.BitrateLast(); } void RTPSender::BitrateUpdated(const BitrateStatistics& stats) { - CriticalSectionScoped cs(statistics_crit_.get()); + uint32_t ssrc; + { + CriticalSectionScoped ssrc_lock(send_critsect_); + ssrc = ssrc_; + } if (bitrate_callback_) { - bitrate_callback_->Notify(stats, ssrc_); + bitrate_callback_->Notify(stats, ssrc); } } + +void RTPSender::SetRtpState(const RtpState& rtp_state) { + SetStartTimestamp(rtp_state.start_timestamp, true); + CriticalSectionScoped lock(send_critsect_); + sequence_number_ = rtp_state.sequence_number; + sequence_number_forced_ = true; + timestamp_ = rtp_state.timestamp; + capture_time_ms_ = rtp_state.capture_time_ms; + last_timestamp_time_ms_ = rtp_state.last_timestamp_time_ms; +} + +RtpState RTPSender::GetRtpState() const { + CriticalSectionScoped lock(send_critsect_); + + RtpState state; + state.sequence_number = sequence_number_; + state.start_timestamp = start_timestamp_; + state.timestamp = timestamp_; + state.capture_time_ms = capture_time_ms_; + state.last_timestamp_time_ms = last_timestamp_time_ms_; + + return state; +} + +void RTPSender::SetRtxRtpState(const RtpState& rtp_state) { + CriticalSectionScoped lock(send_critsect_); + sequence_number_rtx_ = rtp_state.sequence_number; +} + +RtpState RTPSender::GetRtxRtpState() const { + CriticalSectionScoped lock(send_critsect_); + + RtpState state; + state.sequence_number = sequence_number_rtx_; + state.start_timestamp = start_timestamp_; + + return state; +} + } // namespace webrtc diff --git a/modules/rtp_rtcp/source/rtp_sender.h b/modules/rtp_rtcp/source/rtp_sender.h index 291e619b..0cc35cf4 100644 --- a/modules/rtp_rtcp/source/rtp_sender.h +++ b/modules/rtp_rtcp/source/rtp_sender.h @@ -43,12 +43,13 @@ class RTPSenderInterface { virtual uint32_t SSRC() const = 0; virtual uint32_t Timestamp() const = 0; - virtual int32_t BuildRTPheader( - uint8_t *data_buffer, const int8_t payload_type, - const bool marker_bit, const uint32_t capture_time_stamp, - int64_t capture_time_ms, - const bool time_stamp_provided = true, - const bool inc_sequence_number = true) = 0; + virtual int32_t BuildRTPheader(uint8_t* data_buffer, + const int8_t payload_type, + const bool marker_bit, + const uint32_t capture_timestamp, + int64_t capture_time_ms, + const bool timestamp_provided = true, + const bool inc_sequence_number = true) = 0; virtual uint16_t RTPHeaderLength() const = 0; virtual uint16_t IncrementSequenceNumber() = 0; @@ -68,7 +69,8 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer { public: RTPSender(const int32_t id, const bool audio, Clock *clock, Transport *transport, RtpAudioFeedback *audio_feedback, - PacedSender *paced_sender); + PacedSender *paced_sender, + BitrateStatisticsObserver* bitrate_callback); virtual ~RTPSender(); void ProcessBitrate(); @@ -132,13 +134,15 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer { int32_t SetMaxPayloadLength(const uint16_t length, const uint16_t packet_over_head); - int32_t SendOutgoingData( - const FrameType frame_type, const int8_t payload_type, - const uint32_t time_stamp, int64_t capture_time_ms, - const uint8_t *payload_data, const uint32_t payload_size, - const RTPFragmentationHeader *fragmentation, - VideoCodecInformation *codec_info = NULL, - const RTPVideoTypeHeader * rtp_type_hdr = NULL); + int32_t SendOutgoingData(const FrameType frame_type, + const int8_t payload_type, + const uint32_t timestamp, + int64_t capture_time_ms, + const uint8_t* payload_data, + const uint32_t payload_size, + const RTPFragmentationHeader* fragmentation, + VideoCodecInformation* codec_info = NULL, + const RTPVideoTypeHeader* rtp_type_hdr = NULL); // RTP header extension int32_t SetTransmissionTimeOffset( @@ -189,16 +193,19 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer { void RTXStatus(int* mode, uint32_t* ssrc, int* payload_type) const; + uint32_t RtxSsrc() const; void SetRtxSsrc(uint32_t ssrc); void SetRtxPayloadType(int payloadType); // Functions wrapping RTPSenderInterface. virtual int32_t BuildRTPheader( - uint8_t *data_buffer, const int8_t payload_type, - const bool marker_bit, const uint32_t capture_time_stamp, + uint8_t* data_buffer, + const int8_t payload_type, + const bool marker_bit, + const uint32_t capture_timestamp, int64_t capture_time_ms, - const bool time_stamp_provided = true, + const bool timestamp_provided = true, const bool inc_sequence_number = true) OVERRIDE; virtual uint16_t RTPHeaderLength() const OVERRIDE; @@ -269,14 +276,15 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer { void RegisterRtpStatisticsCallback(StreamDataCountersCallback* callback); StreamDataCountersCallback* GetRtpStatisticsCallback() const; - // Called on new send bitrate estimate. - void RegisterBitrateObserver(BitrateStatisticsObserver* observer); - BitrateStatisticsObserver* GetBitrateObserver() const; - uint32_t BitrateSent() const; virtual void BitrateUpdated(const BitrateStatistics& stats) OVERRIDE; + void SetRtpState(const RtpState& rtp_state); + RtpState GetRtpState() const; + void SetRtxRtpState(const RtpState& rtp_state); + RtpState GetRtxRtpState() const; + protected: int32_t CheckPayloadType(const int8_t payload_type, RtpVideoCodecTypes *video_type); @@ -348,7 +356,7 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer { uint16_t packet_over_head_; int8_t payload_type_ GUARDED_BY(send_critsect_); - std::map<int8_t, ModuleRTPUtility::Payload *> payload_type_map_; + std::map<int8_t, RtpUtility::Payload*> payload_type_map_; RtpHeaderExtensionMap rtp_header_extension_map_; int32_t transmission_time_offset_; @@ -363,34 +371,34 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer { // Statistics scoped_ptr<CriticalSectionWrapper> statistics_crit_; - SendDelayMap send_delays_; - std::map<FrameType, uint32_t> frame_counts_; - FrameCountObserver* frame_count_observer_; - StreamDataCounters rtp_stats_; - StreamDataCounters rtx_rtp_stats_; - StreamDataCountersCallback* rtp_stats_callback_; - BitrateStatisticsObserver* bitrate_callback_; + SendDelayMap send_delays_ GUARDED_BY(statistics_crit_); + std::map<FrameType, uint32_t> frame_counts_ GUARDED_BY(statistics_crit_); + FrameCountObserver* frame_count_observer_ GUARDED_BY(statistics_crit_); + StreamDataCounters rtp_stats_ GUARDED_BY(statistics_crit_); + StreamDataCounters rtx_rtp_stats_ GUARDED_BY(statistics_crit_); + StreamDataCountersCallback* rtp_stats_callback_ GUARDED_BY(statistics_crit_); + BitrateStatisticsObserver* const bitrate_callback_; // RTP variables - bool start_time_stamp_forced_; - uint32_t start_time_stamp_; - SSRCDatabase &ssrc_db_; - uint32_t remote_ssrc_; - bool sequence_number_forced_; - uint16_t sequence_number_; - uint16_t sequence_number_rtx_; - bool ssrc_forced_; - uint32_t ssrc_; - uint32_t timestamp_; - int64_t capture_time_ms_; - int64_t last_timestamp_time_ms_; - bool last_packet_marker_bit_; - uint8_t num_csrcs_; - uint32_t csrcs_[kRtpCsrcSize]; - bool include_csrcs_; - int rtx_; - uint32_t ssrc_rtx_; - int payload_type_rtx_; + bool start_timestamp_forced_ GUARDED_BY(send_critsect_); + uint32_t start_timestamp_ GUARDED_BY(send_critsect_); + SSRCDatabase& ssrc_db_ GUARDED_BY(send_critsect_); + uint32_t remote_ssrc_ GUARDED_BY(send_critsect_); + bool sequence_number_forced_ GUARDED_BY(send_critsect_); + uint16_t sequence_number_ GUARDED_BY(send_critsect_); + uint16_t sequence_number_rtx_ GUARDED_BY(send_critsect_); + bool ssrc_forced_ GUARDED_BY(send_critsect_); + uint32_t ssrc_ GUARDED_BY(send_critsect_); + uint32_t timestamp_ GUARDED_BY(send_critsect_); + int64_t capture_time_ms_ GUARDED_BY(send_critsect_); + int64_t last_timestamp_time_ms_ GUARDED_BY(send_critsect_); + bool last_packet_marker_bit_ GUARDED_BY(send_critsect_); + uint8_t num_csrcs_ GUARDED_BY(send_critsect_); + uint32_t csrcs_[kRtpCsrcSize] GUARDED_BY(send_critsect_); + bool include_csrcs_ GUARDED_BY(send_critsect_); + int rtx_ GUARDED_BY(send_critsect_); + uint32_t ssrc_rtx_ GUARDED_BY(send_critsect_); + int payload_type_rtx_ GUARDED_BY(send_critsect_); // Note: Don't access this variable directly, always go through // SetTargetBitrateKbps or GetTargetBitrateKbps. Also remember diff --git a/modules/rtp_rtcp/source/rtp_sender_audio.cc b/modules/rtp_rtcp/source/rtp_sender_audio.cc index 6b3e2276..99c00851 100644 --- a/modules/rtp_rtcp/source/rtp_sender_audio.cc +++ b/modules/rtp_rtcp/source/rtp_sender_audio.cc @@ -89,10 +89,10 @@ int32_t RTPSenderAudio::RegisterAudioPayload( const uint32_t frequency, const uint8_t channels, const uint32_t rate, - ModuleRTPUtility::Payload*& payload) { + RtpUtility::Payload*& payload) { CriticalSectionScoped cs(_sendAudioCritsect); - if (ModuleRTPUtility::StringCompare(payloadName, "cn", 2)) { + if (RtpUtility::StringCompare(payloadName, "cn", 2)) { // we can have multiple CNG payload types if (frequency == 8000) { _cngNBPayloadType = payloadType; @@ -110,14 +110,14 @@ int32_t RTPSenderAudio::RegisterAudioPayload( return -1; } } - if (ModuleRTPUtility::StringCompare(payloadName, "telephone-event", 15)) { + if (RtpUtility::StringCompare(payloadName, "telephone-event", 15)) { // Don't add it to the list // we dont want to allow send with a DTMF payloadtype _dtmfPayloadType = payloadType; return 0; // The default timestamp rate is 8000 Hz, but other rates may be defined. } - payload = new ModuleRTPUtility::Payload; + payload = new RtpUtility::Payload; payload->typeSpecific.Audio.frequency = frequency; payload->typeSpecific.Audio.channels = channels; payload->typeSpecific.Audio.rate = rate; @@ -388,8 +388,8 @@ int32_t RTPSenderAudio::SendAudio( return -1; } uint32_t REDheader = (timestampOffset << 10) + blockLength; - ModuleRTPUtility::AssignUWord24ToBuffer(dataBuffer + rtpHeaderLength, - REDheader); + RtpUtility::AssignUWord24ToBuffer(dataBuffer + rtpHeaderLength, + REDheader); rtpHeaderLength += 3; dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0]; @@ -436,7 +436,7 @@ int32_t RTPSenderAudio::SendAudio( // Update audio level extension, if included. { uint16_t packetSize = payloadSize + rtpHeaderLength; - ModuleRTPUtility::RTPHeaderParser rtp_parser(dataBuffer, packetSize); + RtpUtility::RtpHeaderParser rtp_parser(dataBuffer, packetSize); RTPHeader rtp_header; rtp_parser.Parse(rtp_header); _rtpSender->UpdateAudioLevel(dataBuffer, packetSize, rtp_header, @@ -558,7 +558,7 @@ RTPSenderAudio::SendTelephoneEventPacket(const bool ended, // First byte is Event number, equals key number dtmfbuffer[12] = _dtmfKey; dtmfbuffer[13] = E|R|volume; - ModuleRTPUtility::AssignUWord16ToBuffer(dtmfbuffer+14, duration); + RtpUtility::AssignUWord16ToBuffer(dtmfbuffer + 14, duration); _sendAudioCritsect->Leave(); TRACE_EVENT_INSTANT2("webrtc_rtp", diff --git a/modules/rtp_rtcp/source/rtp_sender_audio.h b/modules/rtp_rtcp/source/rtp_sender_audio.h index 732199c1..d3f67e5e 100644 --- a/modules/rtp_rtcp/source/rtp_sender_audio.h +++ b/modules/rtp_rtcp/source/rtp_sender_audio.h @@ -26,13 +26,12 @@ public: RTPSender* rtpSender); virtual ~RTPSenderAudio(); - int32_t RegisterAudioPayload( - const char payloadName[RTP_PAYLOAD_NAME_SIZE], - const int8_t payloadType, - const uint32_t frequency, - const uint8_t channels, - const uint32_t rate, - ModuleRTPUtility::Payload*& payload); + int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE], + const int8_t payloadType, + const uint32_t frequency, + const uint8_t channels, + const uint32_t rate, + RtpUtility::Payload*& payload); int32_t SendAudio(const FrameType frameType, const int8_t payloadType, diff --git a/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_unittest.cc index 18482890..e08aa202 100644 --- a/modules/rtp_rtcp/source/rtp_sender_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_sender_unittest.cc @@ -94,7 +94,7 @@ class RtpSenderTest : public ::testing::Test { virtual void SetUp() { rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, NULL, - &mock_paced_sender_)); + &mock_paced_sender_, NULL)); rtp_sender_->SetSequenceNumber(kSeqNum); } @@ -204,7 +204,7 @@ TEST_F(RtpSenderTest, BuildRTPPacket) { EXPECT_EQ(kRtpHeaderSize, length); // Verify - webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length); + webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); webrtc::RTPHeader rtp_header; const bool valid_rtp_header = rtp_parser.Parse(rtp_header, NULL); @@ -235,7 +235,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithTransmissionOffsetExtension) { length); // Verify - webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length); + webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); webrtc::RTPHeader rtp_header; RtpHeaderExtensionMap map; @@ -276,7 +276,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithNegativeTransmissionOffsetExtension) { length); // Verify - webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length); + webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); webrtc::RTPHeader rtp_header; RtpHeaderExtensionMap map; @@ -306,7 +306,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithAbsoluteSendTimeExtension) { length); // Verify - webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length); + webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); webrtc::RTPHeader rtp_header; RtpHeaderExtensionMap map; @@ -344,7 +344,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithAudioLevelExtension) { length); // Verify - webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length); + webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); webrtc::RTPHeader rtp_header; // Updating audio level is done in RTPSenderAudio, so simulate it here. @@ -394,7 +394,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithHeaderExtensions) { length); // Verify - webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length); + webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); webrtc::RTPHeader rtp_header; // Updating audio level is done in RTPSenderAudio, so simulate it here. @@ -471,8 +471,8 @@ TEST_F(RtpSenderTest, TrafficSmoothingWithExtensions) { EXPECT_EQ(1, transport_.packets_sent_); EXPECT_EQ(rtp_length, transport_.last_sent_packet_len_); // Parse sent packet. - webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser( - transport_.last_sent_packet_, rtp_length); + webrtc::RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_, + rtp_length); webrtc::RTPHeader rtp_header; RtpHeaderExtensionMap map; map.Register(kRtpExtensionTransmissionTimeOffset, @@ -533,8 +533,8 @@ TEST_F(RtpSenderTest, TrafficSmoothingRetransmits) { EXPECT_EQ(rtp_length, transport_.last_sent_packet_len_); // Parse sent packet. - webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser( - transport_.last_sent_packet_, rtp_length); + webrtc::RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_, + rtp_length); webrtc::RTPHeader rtp_header; RtpHeaderExtensionMap map; map.Register(kRtpExtensionTransmissionTimeOffset, @@ -672,7 +672,7 @@ TEST_F(RtpSenderTest, SendPadding) { TEST_F(RtpSenderTest, SendRedundantPayloads) { MockTransport transport; rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport, NULL, - &mock_paced_sender_)); + &mock_paced_sender_, NULL)); rtp_sender_->SetSequenceNumber(kSeqNum); // Make all packets go through the pacer. EXPECT_CALL(mock_paced_sender_, @@ -744,8 +744,8 @@ TEST_F(RtpSenderTest, SendGenericVideo) { 4321, payload, sizeof(payload), NULL)); - ModuleRTPUtility::RTPHeaderParser rtp_parser(transport_.last_sent_packet_, - transport_.last_sent_packet_len_); + RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_, + transport_.last_sent_packet_len_); webrtc::RTPHeader rtp_header; ASSERT_TRUE(rtp_parser.Parse(rtp_header)); @@ -770,8 +770,8 @@ TEST_F(RtpSenderTest, SendGenericVideo) { 1234, 4321, payload, sizeof(payload), NULL)); - ModuleRTPUtility::RTPHeaderParser rtp_parser2(transport_.last_sent_packet_, - transport_.last_sent_packet_len_); + RtpUtility::RtpHeaderParser rtp_parser2(transport_.last_sent_packet_, + transport_.last_sent_packet_len_); ASSERT_TRUE(rtp_parser.Parse(rtp_header)); payload_data = GetPayloadData(rtp_header, transport_.last_sent_packet_); @@ -865,6 +865,8 @@ TEST_F(RtpSenderTest, BitrateCallbacks) { uint32_t ssrc_; BitrateStatistics bitrate_; } callback; + rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, NULL, + &mock_paced_sender_, &callback)); // Simulate kNumPackets sent with kPacketInterval ms intervals. const uint32_t kNumPackets = 15; @@ -881,8 +883,6 @@ TEST_F(RtpSenderTest, BitrateCallbacks) { rtp_sender_->SetStorePacketsStatus(true, 1); uint32_t ssrc = rtp_sender_->SSRC(); - rtp_sender_->RegisterBitrateObserver(&callback); - // Initial process call so we get a new time window. rtp_sender_->ProcessBitrate(); uint64_t start_time = fake_clock_.CurrentNtpInMilliseconds(); @@ -912,7 +912,7 @@ TEST_F(RtpSenderTest, BitrateCallbacks) { EXPECT_EQ((kPacketOverhead + sizeof(payload)) * 8 * expected_packet_rate, callback.bitrate_.bitrate_bps); - rtp_sender_->RegisterBitrateObserver(NULL); + rtp_sender_.reset(); } class RtpSenderAudioTest : public RtpSenderTest { @@ -922,7 +922,7 @@ class RtpSenderAudioTest : public RtpSenderTest { virtual void SetUp() { payload_ = kAudioPayload; rtp_sender_.reset(new RTPSender(0, true, &fake_clock_, &transport_, NULL, - &mock_paced_sender_)); + &mock_paced_sender_, NULL)); rtp_sender_->SetSequenceNumber(kSeqNum); } }; @@ -1018,8 +1018,8 @@ TEST_F(RtpSenderAudioTest, SendAudio) { 4321, payload, sizeof(payload), NULL)); - ModuleRTPUtility::RTPHeaderParser rtp_parser(transport_.last_sent_packet_, - transport_.last_sent_packet_len_); + RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_, + transport_.last_sent_packet_len_); webrtc::RTPHeader rtp_header; ASSERT_TRUE(rtp_parser.Parse(rtp_header)); @@ -1047,8 +1047,8 @@ TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) { 4321, payload, sizeof(payload), NULL)); - ModuleRTPUtility::RTPHeaderParser rtp_parser(transport_.last_sent_packet_, - transport_.last_sent_packet_len_); + RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_, + transport_.last_sent_packet_len_); webrtc::RTPHeader rtp_header; ASSERT_TRUE(rtp_parser.Parse(rtp_header)); diff --git a/modules/rtp_rtcp/source/rtp_sender_video.cc b/modules/rtp_rtcp/source/rtp_sender_video.cc index 5d8ae166..ea5f7a7e 100644 --- a/modules/rtp_rtcp/source/rtp_sender_video.cc +++ b/modules/rtp_rtcp/source/rtp_sender_video.cc @@ -84,18 +84,20 @@ int32_t RTPSenderVideo::RegisterVideoPayload( const char payloadName[RTP_PAYLOAD_NAME_SIZE], const int8_t payloadType, const uint32_t maxBitRate, - ModuleRTPUtility::Payload*& payload) { + RtpUtility::Payload*& payload) { CriticalSectionScoped cs(_sendVideoCritsect); RtpVideoCodecTypes videoType = kRtpVideoGeneric; - if (ModuleRTPUtility::StringCompare(payloadName, "VP8",3)) { + if (RtpUtility::StringCompare(payloadName, "VP8", 3)) { videoType = kRtpVideoVp8; - } else if (ModuleRTPUtility::StringCompare(payloadName, "I420", 4)) { + } else if (RtpUtility::StringCompare(payloadName, "H264", 4)) { + videoType = kRtpVideoH264; + } else if (RtpUtility::StringCompare(payloadName, "I420", 4)) { videoType = kRtpVideoGeneric; } else { videoType = kRtpVideoGeneric; } - payload = new ModuleRTPUtility::Payload; + payload = new RtpUtility::Payload; payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1); payload->typeSpecific.Video.videoCodecType = videoType; @@ -211,7 +213,7 @@ RTPSenderVideo::SendRTPIntraRequest() data[2] = 0; data[3] = 1; // length - ModuleRTPUtility::AssignUWord32ToBuffer(data+4, _rtpSender.SSRC()); + RtpUtility::AssignUWord32ToBuffer(data + 4, _rtpSender.SSRC()); TRACE_EVENT_INSTANT1("webrtc_rtp", "Video::IntraRequest", diff --git a/modules/rtp_rtcp/source/rtp_sender_video.h b/modules/rtp_rtcp/source/rtp_sender_video.h index daa730e8..82bd1de8 100644 --- a/modules/rtp_rtcp/source/rtp_sender_video.h +++ b/modules/rtp_rtcp/source/rtp_sender_video.h @@ -39,11 +39,10 @@ public: uint16_t FECPacketOverhead() const; - int32_t RegisterVideoPayload( - const char payloadName[RTP_PAYLOAD_NAME_SIZE], - const int8_t payloadType, - const uint32_t maxBitRate, - ModuleRTPUtility::Payload*& payload); + int32_t RegisterVideoPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE], + const int8_t payloadType, + const uint32_t maxBitRate, + RtpUtility::Payload*& payload); int32_t SendVideo(const RtpVideoCodecTypes videoType, const FrameType frameType, diff --git a/modules/rtp_rtcp/source/rtp_utility.cc b/modules/rtp_rtcp/source/rtp_utility.cc index c1f3c642..95389b46 100644 --- a/modules/rtp_rtcp/source/rtp_utility.cc +++ b/modules/rtp_rtcp/source/rtp_utility.cc @@ -66,7 +66,7 @@ ReceiveStatistics* NullObjectReceiveStatistics() { return &null_receive_statistics; } -namespace ModuleRTPUtility { +namespace RtpUtility { enum { kRtcpExpectedVersion = 2, @@ -215,16 +215,16 @@ void RTPPayload::SetType(RtpVideoCodecTypes videoType) { } } -RTPHeaderParser::RTPHeaderParser(const uint8_t* rtpData, - const uint32_t rtpDataLength) - : _ptrRTPDataBegin(rtpData), - _ptrRTPDataEnd(rtpData ? (rtpData + rtpDataLength) : NULL) { +RtpHeaderParser::RtpHeaderParser(const uint8_t* rtpData, + const size_t rtpDataLength) + : _ptrRTPDataBegin(rtpData), + _ptrRTPDataEnd(rtpData ? (rtpData + rtpDataLength) : NULL) { } -RTPHeaderParser::~RTPHeaderParser() { +RtpHeaderParser::~RtpHeaderParser() { } -bool RTPHeaderParser::RTCP() const { +bool RtpHeaderParser::RTCP() const { // 72 to 76 is reserved for RTP // 77 to 79 is not reserver but they are not assigned we will block them // for RTCP 200 SR == marker bit + 72 @@ -299,7 +299,7 @@ bool RTPHeaderParser::RTCP() const { return RTCP; } -bool RTPHeaderParser::ParseRtcp(RTPHeader* header) const { +bool RtpHeaderParser::ParseRtcp(RTPHeader* header) const { assert(header != NULL); const ptrdiff_t length = _ptrRTPDataEnd - _ptrRTPDataBegin; @@ -328,7 +328,7 @@ bool RTPHeaderParser::ParseRtcp(RTPHeader* header) const { return true; } -bool RTPHeaderParser::Parse(RTPHeader& header, +bool RtpHeaderParser::Parse(RTPHeader& header, RtpHeaderExtensionMap* ptrExtensionMap) const { const ptrdiff_t length = _ptrRTPDataEnd - _ptrRTPDataBegin; if (length < kRtpMinParseLength) { @@ -441,7 +441,7 @@ bool RTPHeaderParser::Parse(RTPHeader& header, return true; } -void RTPHeaderParser::ParseOneByteExtensionHeader( +void RtpHeaderParser::ParseOneByteExtensionHeader( RTPHeader& header, const RtpHeaderExtensionMap* ptrExtensionMap, const uint8_t* ptrRTPDataExtensionEnd, @@ -552,10 +552,9 @@ void RTPHeaderParser::ParseOneByteExtensionHeader( } } -uint8_t RTPHeaderParser::ParsePaddingBytes( - const uint8_t* ptrRTPDataExtensionEnd, - const uint8_t* ptr) const { - +uint8_t RtpHeaderParser::ParsePaddingBytes( + const uint8_t* ptrRTPDataExtensionEnd, + const uint8_t* ptr) const { uint8_t num_zero_bytes = 0; while (ptrRTPDataExtensionEnd - ptr > 0) { if (*ptr != 0) { @@ -768,6 +767,6 @@ int RTPPayloadParser::ParseVP8TIDAndKeyIdx(RTPPayloadVP8* vp8, return 0; } -} // namespace ModuleRTPUtility +} // namespace RtpUtility } // namespace webrtc diff --git a/modules/rtp_rtcp/source/rtp_utility.h b/modules/rtp_rtcp/source/rtp_utility.h index 732301f6..ef50570d 100644 --- a/modules/rtp_rtcp/source/rtp_utility.h +++ b/modules/rtp_rtcp/source/rtp_utility.h @@ -28,8 +28,7 @@ RtpFeedback* NullObjectRtpFeedback(); RtpAudioFeedback* NullObjectRtpAudioFeedback(); ReceiveStatistics* NullObjectReceiveStatistics(); -namespace ModuleRTPUtility -{ +namespace RtpUtility { // January 1970, in NTP seconds. const uint32_t NTP_JAN_1970 = 2208988800UL; @@ -92,12 +91,10 @@ namespace ModuleRTPUtility */ uint32_t BufferToUWord32(const uint8_t* dataBuffer); - class RTPHeaderParser - { + class RtpHeaderParser { public: - RTPHeaderParser(const uint8_t* rtpData, - const uint32_t rtpDataLength); - ~RTPHeaderParser(); + RtpHeaderParser(const uint8_t* rtpData, size_t rtpDataLength); + ~RtpHeaderParser(); bool RTCP() const; bool ParseRtcp(RTPHeader* header) const; @@ -207,7 +204,7 @@ namespace ModuleRTPUtility const RtpVideoCodecTypes _videoType; }; -} // namespace ModuleRTPUtility + } // namespace RtpUtility } // namespace webrtc diff --git a/modules/rtp_rtcp/source/rtp_utility_unittest.cc b/modules/rtp_rtcp/source/rtp_utility_unittest.cc index d33eaf4c..27170d55 100644 --- a/modules/rtp_rtcp/source/rtp_utility_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_utility_unittest.cc @@ -8,9 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ - /* - * This file conatins unit tests for the ModuleRTPUtility. + * This file conatins unit tests for the RtpUtility. */ #include "testing/gtest/include/gtest/gtest.h" @@ -20,9 +19,9 @@ namespace webrtc { -using ModuleRTPUtility::RTPPayloadParser; -using ModuleRTPUtility::RTPPayload; -using ModuleRTPUtility::RTPPayloadVP8; +using RtpUtility::RTPPayloadParser; +using RtpUtility::RTPPayload; +using RtpUtility::RTPPayloadVP8; // Payload descriptor // 0 1 2 3 4 5 6 7 @@ -81,7 +80,7 @@ TEST(ParseVP8Test, BasicHeader) { RTPPayload parsedPacket; ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket)); - EXPECT_EQ(ModuleRTPUtility::kPFrame, parsedPacket.frameType); + EXPECT_EQ(RtpUtility::kPFrame, parsedPacket.frameType); EXPECT_EQ(kRtpVideoVp8, parsedPacket.type); VerifyBasicHeader(parsedPacket.info.VP8, 0 /*N*/, 1 /*S*/, 4 /*PartID*/); @@ -102,7 +101,7 @@ TEST(ParseVP8Test, PictureID) { RTPPayload parsedPacket; ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket)); - EXPECT_EQ(ModuleRTPUtility::kPFrame, parsedPacket.frameType); + EXPECT_EQ(RtpUtility::kPFrame, parsedPacket.frameType); EXPECT_EQ(kRtpVideoVp8, parsedPacket.type); VerifyBasicHeader(parsedPacket.info.VP8, 1 /*N*/, 0 /*S*/, 0 /*PartID*/); @@ -141,7 +140,7 @@ TEST(ParseVP8Test, Tl0PicIdx) { RTPPayload parsedPacket; ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket)); - EXPECT_EQ(ModuleRTPUtility::kIFrame, parsedPacket.frameType); + EXPECT_EQ(RtpUtility::kIFrame, parsedPacket.frameType); EXPECT_EQ(kRtpVideoVp8, parsedPacket.type); VerifyBasicHeader(parsedPacket.info.VP8, 0 /*N*/, 1 /*S*/, 0 /*PartID*/); @@ -164,7 +163,7 @@ TEST(ParseVP8Test, TIDAndLayerSync) { RTPPayload parsedPacket; ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket)); - EXPECT_EQ(ModuleRTPUtility::kPFrame, parsedPacket.frameType); + EXPECT_EQ(RtpUtility::kPFrame, parsedPacket.frameType); EXPECT_EQ(kRtpVideoVp8, parsedPacket.type); VerifyBasicHeader(parsedPacket.info.VP8, 0 /*N*/, 0 /*S*/, 8 /*PartID*/); @@ -188,7 +187,7 @@ TEST(ParseVP8Test, KeyIdx) { RTPPayload parsedPacket; ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket)); - EXPECT_EQ(ModuleRTPUtility::kPFrame, parsedPacket.frameType); + EXPECT_EQ(RtpUtility::kPFrame, parsedPacket.frameType); EXPECT_EQ(kRtpVideoVp8, parsedPacket.type); VerifyBasicHeader(parsedPacket.info.VP8, 0 /*N*/, 0 /*S*/, 8 /*PartID*/); @@ -214,7 +213,7 @@ TEST(ParseVP8Test, MultipleExtensions) { RTPPayload parsedPacket; ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket)); - EXPECT_EQ(ModuleRTPUtility::kPFrame, parsedPacket.frameType); + EXPECT_EQ(RtpUtility::kPFrame, parsedPacket.frameType); EXPECT_EQ(kRtpVideoVp8, parsedPacket.type); VerifyBasicHeader(parsedPacket.info.VP8, 0 /*N*/, 0 /*S*/, 8 /*PartID*/); @@ -263,7 +262,7 @@ TEST(ParseVP8Test, TestWithPacketizer) { RTPPayload parsedPacket; ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket)); - EXPECT_EQ(ModuleRTPUtility::kIFrame, parsedPacket.frameType); + EXPECT_EQ(RtpUtility::kIFrame, parsedPacket.frameType); EXPECT_EQ(kRtpVideoVp8, parsedPacket.type); VerifyBasicHeader(parsedPacket.info.VP8, diff --git a/modules/rtp_rtcp/test/testAPI/test_api.h b/modules/rtp_rtcp/test/testAPI/test_api.h index 8061ce01..1c6b8838 100644 --- a/modules/rtp_rtcp/test/testAPI/test_api.h +++ b/modules/rtp_rtcp/test/testAPI/test_api.h @@ -52,7 +52,9 @@ class LoopBackTransport : public webrtc::Transport { } RTPHeader header; scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create()); - if (!parser->Parse(static_cast<const uint8_t*>(data), len, &header)) { + if (!parser->Parse(static_cast<const uint8_t*>(data), + static_cast<size_t>(len), + &header)) { return -1; } PayloadUnion payload_specific; diff --git a/modules/rtp_rtcp/test/testAPI/test_api_video.cc b/modules/rtp_rtcp/test/testAPI/test_api_video.cc index 94d1e52e..4c4944d9 100644 --- a/modules/rtp_rtcp/test/testAPI/test_api_video.cc +++ b/modules/rtp_rtcp/test/testAPI/test_api_video.cc @@ -83,11 +83,9 @@ class RtpRtcpVideoTest : public ::testing::Test { uint32_t sequence_number) { dataBuffer[0] = static_cast<uint8_t>(0x80); // version 2 dataBuffer[1] = static_cast<uint8_t>(kPayloadType); - ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer + 2, - sequence_number); - ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer + 4, timestamp); - ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer + 8, - 0x1234); // SSRC. + RtpUtility::AssignUWord16ToBuffer(dataBuffer + 2, sequence_number); + RtpUtility::AssignUWord32ToBuffer(dataBuffer + 4, timestamp); + RtpUtility::AssignUWord32ToBuffer(dataBuffer + 8, 0x1234); // SSRC. int32_t rtpHeaderLength = 12; return rtpHeaderLength; } diff --git a/modules/rtp_rtcp/test/testFec/test_fec.cc b/modules/rtp_rtcp/test/testFec/test_fec.cc index fc11fbee..bdb53afe 100644 --- a/modules/rtp_rtcp/test/testFec/test_fec.cc +++ b/modules/rtp_rtcp/test/testFec/test_fec.cc @@ -134,7 +134,7 @@ TEST(FecTest, FecTest) { fclose(randomSeedFile); randomSeedFile = NULL; - uint16_t seqNum = static_cast<uint16_t>(rand()); + uint16_t seqNum = 0; uint32_t timeStamp = static_cast<uint32_t>(rand()); const uint32_t ssrc = static_cast<uint32_t>(rand()); @@ -224,6 +224,11 @@ TEST(FecTest, FecTest) { } // Construct media packets. + // Reset the sequence number here for each FEC code/mask tested + // below, to avoid sequence number wrap-around. In actual decoding, + // old FEC packets in list are dropped if sequence number wrap + // around is detected. This case is currently not handled below. + seqNum = 0; for (uint32_t i = 0; i < numMediaPackets; ++i) { mediaPacket = new ForwardErrorCorrection::Packet; mediaPacketList.push_back(mediaPacket); @@ -254,12 +259,10 @@ TEST(FecTest, FecTest) { // Only push one (fake) frame to the FEC. mediaPacket->data[1] &= 0x7f; - ModuleRTPUtility::AssignUWord16ToBuffer(&mediaPacket->data[2], - seqNum); - ModuleRTPUtility::AssignUWord32ToBuffer(&mediaPacket->data[4], - timeStamp); - ModuleRTPUtility::AssignUWord32ToBuffer(&mediaPacket->data[8], - ssrc); + RtpUtility::AssignUWord16ToBuffer(&mediaPacket->data[2], seqNum); + RtpUtility::AssignUWord32ToBuffer(&mediaPacket->data[4], + timeStamp); + RtpUtility::AssignUWord32ToBuffer(&mediaPacket->data[8], ssrc); // Generate random values for payload for (int32_t j = 12; j < mediaPacket->length; ++j) { mediaPacket->data[j] = static_cast<uint8_t>(rand() % 256); @@ -298,7 +301,7 @@ TEST(FecTest, FecTest) { memcpy(receivedPacket->pkt->data, mediaPacket->data, mediaPacket->length); receivedPacket->seq_num = - ModuleRTPUtility::BufferToUWord16(&mediaPacket->data[2]); + RtpUtility::BufferToUWord16(&mediaPacket->data[2]); receivedPacket->is_fec = false; } mediaPacketIdx++; diff --git a/modules/utility/source/rtp_dump_impl.h b/modules/utility/source/rtp_dump_impl.h index 04ae7dfc..ff3f07ce 100644 --- a/modules/utility/source/rtp_dump_impl.h +++ b/modules/utility/source/rtp_dump_impl.h @@ -35,7 +35,7 @@ private: inline uint16_t RtpDumpHtons(uint16_t x) const; // Return true if the packet starts with a valid RTCP header. - // Note: See ModuleRTPUtility::RTPHeaderParser::RTCP() for details on how + // Note: See RtpUtility::RtpHeaderParser::RTCP() for details on how // to determine if the packet is an RTCP packet. bool RTCP(const uint8_t* packet) const; diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn index f5a69b18..0dc6721b 100644 --- a/modules/video_coding/BUILD.gn +++ b/modules/video_coding/BUILD.gn @@ -74,13 +74,49 @@ source_set("video_coding") { } source_set("video_coding_utility") { - # TODO(stefan): Implement. + sources = [ + "utility/exp_filter.cc", + "utility/include/exp_filter.h", + "utility/include/frame_dropper.h", + "utility/frame_dropper.cc", + ] + + deps = [ "../../system_wrappers" ] } source_set("webrtc_i420") { - # TODO(stefan): Implement. + sources = [ + "codecs/i420/main/source/i420.cc", + "codecs/i420/main/interface/i420.h", + ] + + deps = [ "../../system_wrappers" ] } +# TODO(holmer): Some files below has been commented out since libvpx is still +# missing a BUILD.gn file. source_set("webrtc_vp8") { - # TODO(stefan): Implement. + sources = [ +# "codecs/vp8/default_temporal_layers.cc", +# "codecs/vp8/default_temporal_layers.h", +# "codecs/vp8/realtime_temporal_layers.cc", +# "codecs/vp8/reference_picture_selection.cc", +# "codecs/vp8/reference_picture_selection.h", + "codecs/vp8/include/vp8.h", + "codecs/vp8/include/vp8_common_types.h", + "codecs/vp8/temporal_layers.h", + "codecs/vp8/vp8_factory.cc", +# "codecs/vp8/vp8_impl.cc", +# "codecs/vp8/vp8_impl.h", + ] + deps = [ + ":video_coding_utility", + "../../common_video", + "../../system_wrappers", + ] +# if (build_libvpx) { +# deps += [ +# "//third_party/libvpx", +# ] +# } } diff --git a/modules/video_coding/main/interface/video_coding_defines.h b/modules/video_coding/main/interface/video_coding_defines.h index fab91afd..c5f93cb1 100644 --- a/modules/video_coding/main/interface/video_coding_defines.h +++ b/modules/video_coding/main/interface/video_coding_defines.h @@ -40,6 +40,7 @@ namespace webrtc { #define VCM_ULPFEC_PAYLOAD_TYPE 97 #define VCM_VP8_PAYLOAD_TYPE 100 #define VCM_I420_PAYLOAD_TYPE 124 +#define VCM_H264_PAYLOAD_TYPE 127 enum VCMVideoProtection { kProtectionNack, // Both send-side and receive-side diff --git a/modules/video_coding/main/source/codec_database.cc b/modules/video_coding/main/source/codec_database.cc index e7a9d91b..e5013689 100644 --- a/modules/video_coding/main/source/codec_database.cc +++ b/modules/video_coding/main/source/codec_database.cc @@ -102,6 +102,30 @@ bool VCMCodecDataBase::Codec(int list_id, return true; } #endif +#ifdef VIDEOCODEC_H264 + case VCM_H264_IDX: { + strncpy(settings->plName, "H264", 5); + settings->codecType = kVideoCodecH264; + // 96 to 127 dynamic payload types for video codecs. + settings->plType = VCM_H264_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->qpMax = 56; + settings->codecSpecific.H264.profile = kProfileBase; + settings->codecSpecific.H264.frameDroppingOn = true; + settings->codecSpecific.H264.keyFrameInterval = 3000; + settings->codecSpecific.H264.spsData = NULL; + settings->codecSpecific.H264.spsLen = 0; + settings->codecSpecific.H264.ppsData = NULL; + settings->codecSpecific.H264.ppsLen = 0; + return true; + } +#endif #ifdef VIDEOCODEC_I420 case VCM_I420_IDX: { strncpy(settings->plName, "I420", 5); @@ -316,8 +340,14 @@ bool VCMCodecDataBase::RequiresEncoderReset(const VideoCodec& new_send_codec) { case kVideoCodecVP8: if (memcmp(&new_send_codec.codecSpecific.VP8, &send_codec_.codecSpecific.VP8, - sizeof(new_send_codec.codecSpecific.VP8)) != - 0) { + sizeof(new_send_codec.codecSpecific.VP8)) != 0) { + return true; + } + break; + case kVideoCodecH264: + if (memcmp(&new_send_codec.codecSpecific.H264, + &send_codec_.codecSpecific.H264, + sizeof(new_send_codec.codecSpecific.H264)) != 0) { return true; } break; @@ -619,6 +649,7 @@ VCMGenericDecoder* VCMCodecDataBase::CreateDecoder(VideoCodecType type) const { return new VCMGenericDecoder(*(new I420Decoder)); #endif default: + LOG(LS_WARNING) << "No internal decoder of this type exists."; return NULL; } } diff --git a/modules/video_coding/main/source/internal_defines.h b/modules/video_coding/main/source/internal_defines.h index efc6d8dd..ef42c628 100644 --- a/modules/video_coding/main/source/internal_defines.h +++ b/modules/video_coding/main/source/internal_defines.h @@ -35,16 +35,21 @@ inline uint32_t MaskWord64ToUWord32(int64_t w64) // 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 + #define VCM_VP8_IDX (VCM_NO_CODEC_IDX + 1) #else #define VCM_VP8_IDX VCM_NO_CODEC_IDX #endif +#ifdef VIDEOCODEC_H264 + #define VCM_H264_IDX (VCM_VP8_IDX + 1) +#else + #define VCM_H264_IDX VCM_VP8_IDX +#endif #ifdef VIDEOCODEC_I420 - #define VCM_I420_IDX VCM_VP8_IDX + 1 + #define VCM_I420_IDX (VCM_H264_IDX + 1) #else - #define VCM_I420_IDX VCM_VP8_IDX + #define VCM_I420_IDX VCM_H264_IDX #endif -#define VCM_NUM_VIDEO_CODECS_AVAILABLE VCM_I420_IDX + 1 +#define VCM_NUM_VIDEO_CODECS_AVAILABLE (VCM_I420_IDX + 1) #define VCM_NO_RECEIVER_ID 0 diff --git a/modules/video_coding/main/source/video_coding_impl.h b/modules/video_coding/main/source/video_coding_impl.h index bf0bc790..816552fe 100644 --- a/modules/video_coding/main/source/video_coding_impl.h +++ b/modules/video_coding/main/source/video_coding_impl.h @@ -25,6 +25,7 @@ #include "webrtc/modules/video_coding/main/source/timing.h" #include "webrtc/system_wrappers/interface/clock.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" +#include "webrtc/system_wrappers/interface/thread_annotations.h" namespace webrtc { @@ -199,21 +200,25 @@ class VideoReceiver { // in any frame }; - Clock* clock_; + Clock* const clock_; scoped_ptr<CriticalSectionWrapper> process_crit_sect_; CriticalSectionWrapper* _receiveCritSect; - bool _receiverInited; + bool _receiverInited GUARDED_BY(_receiveCritSect); VCMTiming _timing; VCMTiming _dualTiming; VCMReceiver _receiver; VCMReceiver _dualReceiver; VCMDecodedFrameCallback _decodedFrameCallback; VCMDecodedFrameCallback _dualDecodedFrameCallback; - VCMFrameTypeCallback* _frameTypeCallback; - VCMReceiveStatisticsCallback* _receiveStatsCallback; - VCMDecoderTimingCallback* _decoderTimingCallback; - VCMPacketRequestCallback* _packetRequestCallback; - VCMRenderBufferSizeCallback* render_buffer_callback_; + VCMFrameTypeCallback* _frameTypeCallback GUARDED_BY(process_crit_sect_); + VCMReceiveStatisticsCallback* _receiveStatsCallback + GUARDED_BY(process_crit_sect_); + VCMDecoderTimingCallback* _decoderTimingCallback + GUARDED_BY(process_crit_sect_); + VCMPacketRequestCallback* _packetRequestCallback + GUARDED_BY(process_crit_sect_); + VCMRenderBufferSizeCallback* render_buffer_callback_ + GUARDED_BY(process_crit_sect_); VCMGenericDecoder* _decoder; VCMGenericDecoder* _dualDecoder; #ifdef DEBUG_DECODER_BIT_STREAM @@ -221,9 +226,9 @@ class VideoReceiver { #endif VCMFrameBuffer _frameFromFile; VCMKeyRequestMode _keyRequestMode; - bool _scheduleKeyRequest; - size_t max_nack_list_size_; - EncodedImageCallback* pre_decode_image_callback_; + bool _scheduleKeyRequest GUARDED_BY(process_crit_sect_); + size_t max_nack_list_size_ GUARDED_BY(process_crit_sect_); + EncodedImageCallback* pre_decode_image_callback_ GUARDED_BY(_receiveCritSect); VCMCodecDataBase _codecDataBase; VCMProcessTimer _receiveStatsTimer; diff --git a/modules/video_coding/main/source/video_receiver.cc b/modules/video_coding/main/source/video_receiver.cc index 5bc1c90f..0b561249 100644 --- a/modules/video_coding/main/source/video_receiver.cc +++ b/modules/video_coding/main/source/video_receiver.cc @@ -271,8 +271,6 @@ int32_t VideoReceiver::SetVideoProtection(VCMVideoProtection videoProtection, // Initialize receiver, resets codec database etc int32_t VideoReceiver::InitializeReceiver() { - CriticalSectionScoped receive_cs(_receiveCritSect); - CriticalSectionScoped process_cs(process_crit_sect_.get()); int32_t ret = _receiver.Initialize(); if (ret < 0) { return ret; @@ -285,15 +283,22 @@ int32_t VideoReceiver::InitializeReceiver() { _codecDataBase.ResetReceiver(); _timing.Reset(); - _decoder = NULL; - _decodedFrameCallback.SetUserReceiveCallback(NULL); - _receiverInited = true; - _frameTypeCallback = NULL; - _receiveStatsCallback = NULL; - _decoderTimingCallback = NULL; - _packetRequestCallback = NULL; - _keyRequestMode = kKeyOnError; - _scheduleKeyRequest = false; + { + CriticalSectionScoped receive_cs(_receiveCritSect); + _receiverInited = true; + } + + { + CriticalSectionScoped process_cs(process_crit_sect_.get()); + _decoder = NULL; + _decodedFrameCallback.SetUserReceiveCallback(NULL); + _frameTypeCallback = NULL; + _receiveStatsCallback = NULL; + _decoderTimingCallback = NULL; + _packetRequestCallback = NULL; + _keyRequestMode = kKeyOnError; + _scheduleKeyRequest = false; + } return VCM_OK; } @@ -781,7 +786,6 @@ void VideoReceiver::SetNackSettings(size_t max_nack_list_size, int max_packet_age_to_nack, int max_incomplete_time_ms) { if (max_nack_list_size != 0) { - CriticalSectionScoped receive_cs(_receiveCritSect); CriticalSectionScoped process_cs(process_crit_sect_.get()); max_nack_list_size_ = max_nack_list_size; } diff --git a/modules/video_coding/main/test/pcap_file_reader.cc b/modules/video_coding/main/test/pcap_file_reader.cc index 3d4e2659..68c85665 100644 --- a/modules/video_coding/main/test/pcap_file_reader.cc +++ b/modules/video_coding/main/test/pcap_file_reader.cc @@ -268,8 +268,7 @@ class PcapFileReaderImpl : public RtpPacketSourceInterface { } TRY(Read(read_buffer_, marker.payload_length)); - ModuleRTPUtility::RTPHeaderParser rtp_parser(read_buffer_, - marker.payload_length); + RtpUtility::RtpHeaderParser rtp_parser(read_buffer_, marker.payload_length); if (rtp_parser.RTCP()) { rtp_parser.ParseRtcp(&marker.rtp_header); packets_.push_back(marker); diff --git a/modules/video_coding/main/test/pcap_file_reader_unittest.cc b/modules/video_coding/main/test/pcap_file_reader_unittest.cc index 18100714..c6f1d511 100644 --- a/modules/video_coding/main/test/pcap_file_reader_unittest.cc +++ b/modules/video_coding/main/test/pcap_file_reader_unittest.cc @@ -55,7 +55,7 @@ class TestPcapFileReader : public ::testing::Test { EXPECT_GE(kBufferSize, length); length = kBufferSize; - ModuleRTPUtility::RTPHeaderParser rtp_header_parser(data, length); + RtpUtility::RtpHeaderParser rtp_header_parser(data, length); webrtc::RTPHeader header; if (!rtp_header_parser.RTCP() && rtp_header_parser.Parse(header, NULL)) { pps[header.ssrc]++; diff --git a/modules/video_processing/BUILD.gn b/modules/video_processing/BUILD.gn index 40171caf..724a9dcb 100644 --- a/modules/video_processing/BUILD.gn +++ b/modules/video_processing/BUILD.gn @@ -8,6 +8,51 @@ import("../../build/webrtc.gni") +build_video_processing_sse2 = cpu_arch == "x86" || cpu_arch == "x64" + source_set("video_processing") { - # TODO(stefan): Implement. + sources = [ + "main/interface/video_processing.h", + "main/interface/video_processing_defines.h", + "main/source/brighten.cc", + "main/source/brighten.h", + "main/source/brightness_detection.cc", + "main/source/brightness_detection.h", + "main/source/color_enhancement.cc", + "main/source/color_enhancement.h", + "main/source/color_enhancement_private.h", + "main/source/content_analysis.cc", + "main/source/content_analysis.h", + "main/source/deflickering.cc", + "main/source/deflickering.h", + "main/source/denoising.cc", + "main/source/denoising.h", + "main/source/frame_preprocessor.cc", + "main/source/frame_preprocessor.h", + "main/source/spatial_resampler.cc", + "main/source/spatial_resampler.h", + "main/source/video_decimator.cc", + "main/source/video_decimator.h", + "main/source/video_processing_impl.cc", + "main/source/video_processing_impl.h", + ] + + deps = [ + "../../common_audio", + "../../common_video", + "../../modules/utility", + "../../system_wrappers", + ] + if (build_video_processing_sse2) { + deps += [ ":video_processing_sse2" ] + } +} + +if (build_video_processing_sse2) { + source_set("video_processing_sse2") { + sources = [ "main/source/content_analysis_sse2.cc" ] + if (is_posix) { + cflags = [ "-msse2" ] + } + } } diff --git a/system_wrappers/source/condition_variable_unittest.cc b/system_wrappers/source/condition_variable_unittest.cc index 0d287b71..b5c1c360 100644 --- a/system_wrappers/source/condition_variable_unittest.cc +++ b/system_wrappers/source/condition_variable_unittest.cc @@ -14,13 +14,11 @@ #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/thread_wrapper.h" #include "webrtc/system_wrappers/interface/trace.h" -#include "webrtc/system_wrappers/source/unittest_utilities.h" namespace webrtc { namespace { -const int kLogTrace = false; // Set to true to enable debug logging to stdout. const int kLongWaitMs = 100 * 1000; // A long time in testing terms const int kShortWaitMs = 2 * 1000; // Long enough for process switches to happen @@ -143,9 +141,7 @@ bool WaitingRunFunction(void* obj) { class CondVarTest : public ::testing::Test { public: - CondVarTest() - : trace_(kLogTrace) { - } + CondVarTest() {} virtual void SetUp() { thread_ = ThreadWrapper::CreateThread(&WaitingRunFunction, @@ -171,7 +167,6 @@ class CondVarTest : public ::testing::Test { Baton baton_; private: - ScopedTracing trace_; ThreadWrapper* thread_; }; diff --git a/system_wrappers/source/critical_section_unittest.cc b/system_wrappers/source/critical_section_unittest.cc index 5c416b2d..9b88e6b3 100644 --- a/system_wrappers/source/critical_section_unittest.cc +++ b/system_wrappers/source/critical_section_unittest.cc @@ -14,14 +14,11 @@ #include "webrtc/system_wrappers/interface/sleep.h" #include "webrtc/system_wrappers/interface/thread_wrapper.h" #include "webrtc/system_wrappers/interface/trace.h" -#include "webrtc/system_wrappers/source/unittest_utilities.h" namespace webrtc { namespace { -const bool kLogTrace = false; // Set to true to enable debug logging to stdout. - // Cause a process switch. Needed to avoid depending on // busy-wait in tests. static void SwitchProcess() { @@ -54,8 +51,7 @@ private: class CritSectTest : public ::testing::Test { public: - CritSectTest() : trace_(kLogTrace) { - } + CritSectTest() {} // Waits a number of cycles for the count to reach a given value. // Returns true if the target is reached or passed. @@ -70,9 +66,6 @@ public: } return (count->Count() >= target); } - -private: - ScopedTracing trace_; }; bool LockUnlockThenStopRunFunction(void* obj) { diff --git a/system_wrappers/source/logging_unittest.cc b/system_wrappers/source/logging_unittest.cc index 19f13940..6e45c5c6 100644 --- a/system_wrappers/source/logging_unittest.cc +++ b/system_wrappers/source/logging_unittest.cc @@ -46,22 +46,19 @@ class LoggingTest : public ::testing::Test, public TraceCallback { void SetUp() { Trace::CreateTrace(); Trace::SetTraceCallback(this); - // Reduce the chance that spurious traces will ruin the test. - Trace::set_level_filter(kTraceWarning | kTraceError); } void TearDown() { - CriticalSectionScoped cs(crit_.get()); Trace::SetTraceCallback(NULL); Trace::ReturnTrace(); + CriticalSectionScoped cs(crit_.get()); ASSERT_EQ(kTraceNone, level_) << "Print() was not called"; } scoped_ptr<CriticalSectionWrapper> crit_; scoped_ptr<ConditionVariableWrapper> cv_; - TraceLevel level_; - int length_; - std::ostringstream expected_log_; + TraceLevel level_ GUARDED_BY(crit_); + std::ostringstream expected_log_ GUARDED_BY(crit_); }; TEST_F(LoggingTest, LogStream) { diff --git a/system_wrappers/source/system_wrappers_tests.gyp b/system_wrappers/source/system_wrappers_tests.gyp index f2f61566..3d08c0d0 100644 --- a/system_wrappers/source/system_wrappers_tests.gyp +++ b/system_wrappers/source/system_wrappers_tests.gyp @@ -35,7 +35,6 @@ 'stl_util_unittest.cc', 'thread_unittest.cc', 'thread_posix_unittest.cc', - 'unittest_utilities_unittest.cc', ], 'conditions': [ ['enable_data_logging==1', { diff --git a/system_wrappers/source/unittest_utilities.h b/system_wrappers/source/unittest_utilities.h deleted file mode 100644 index b32308e6..00000000 --- a/system_wrappers/source/unittest_utilities.h +++ /dev/null @@ -1,82 +0,0 @@ -/* - * 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_SYSTEM_WRAPPERS_SOURCE_UNITTEST_UTILITIES_H_ -#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_UNITTEST_UTILITIES_H_ - -// This file contains utilities that make it simpler to write unittests -// that are appropriate for the system_wrappers classes. - -#include <stdio.h> -#include <string.h> - -#include "webrtc/system_wrappers/interface/trace.h" - -namespace webrtc { - -class TestTraceCallback : public TraceCallback { - public: - virtual void Print(TraceLevel level, const char* msg, int length) { - if (msg) { - char* cmd_print = new char[length+1]; - memcpy(cmd_print, msg, length); - cmd_print[length] = '\0'; - printf("%s\n", cmd_print); - fflush(stdout); - delete[] cmd_print; - } - } -}; - -// A class that turns on tracing to stdout at the beginning of the test, -// and turns it off once the test is finished. -// Intended usage: -// class SomeTest : public ::testing::Test { -// protected: -// SomeTest() -// : trace_(false) {} // Change to true to turn on tracing. -// private: -// ScopedTracing trace_; -// } -class ScopedTracing { - public: - explicit ScopedTracing(bool logOn) { - logging_ = logOn; - StartTrace(); - } - - ~ScopedTracing() { - StopTrace(); - } - - private: - void StartTrace() { - if (logging_) { - Trace::CreateTrace(); - Trace::set_level_filter(webrtc::kTraceAll); - Trace::SetTraceCallback(&trace_); - } - } - - void StopTrace() { - if (logging_) { - Trace::SetTraceCallback(NULL); - Trace::ReturnTrace(); - } - } - - private: - bool logging_; - TestTraceCallback trace_; -}; - -} // namespace webrtc - -#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_UNITTEST_UTILITIES_H_ diff --git a/system_wrappers/source/unittest_utilities_unittest.cc b/system_wrappers/source/unittest_utilities_unittest.cc deleted file mode 100644 index be6647b7..00000000 --- a/system_wrappers/source/unittest_utilities_unittest.cc +++ /dev/null @@ -1,34 +0,0 @@ -/* - * 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 "webrtc/system_wrappers/source/unittest_utilities.h" - -#include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/system_wrappers/interface/trace.h" - -namespace webrtc { - -// These tests merely check that the code compiles and that no -// fatal accidents happen when logging. -TEST(UnittestUtilities, TraceOn) { - ScopedTracing trace(true); - WEBRTC_TRACE(kTraceInfo, kTraceUtility, 0, "Log line that should appear"); - // TODO(hta): Verify that output appears. - // Note - output is written on another thread, so can take time to appear. -} - -TEST(UnittestUtilities, TraceOff) { - ScopedTracing trace(false); - WEBRTC_TRACE(kTraceInfo, kTraceUtility, 0, - "Log line that should not appear"); - // TODO(hta): Verify that no output appears. -} - -} // namespace webrtc diff --git a/test/OWNERS b/test/OWNERS index 1532e3fd..98a0d4d1 100644 --- a/test/OWNERS +++ b/test/OWNERS @@ -1,5 +1,6 @@ -phoglund@webrtc.org kjellander@webrtc.org +pbos@webrtc.org +phoglund@webrtc.org per-file *.isolate=kjellander@webrtc.org diff --git a/test/call_test.cc b/test/call_test.cc index 5d8b8e67..16dad719 100644 --- a/test/call_test.cc +++ b/test/call_test.cc @@ -15,8 +15,9 @@ namespace webrtc { namespace test { CallTest::CallTest() - : send_stream_(NULL), - fake_encoder_(Clock::GetRealTimeClock()) { + : clock_(Clock::GetRealTimeClock()), + send_stream_(NULL), + fake_encoder_(clock_) { } CallTest::~CallTest() { } @@ -85,7 +86,7 @@ void CallTest::CreateReceiverCall(const Call::Config& config) { void CallTest::CreateSendConfig(size_t num_streams) { assert(num_streams <= kNumSsrcs); - send_config_ = sender_call_->GetDefaultSendConfig(); + send_config_ = VideoSendStream::Config(); send_config_.encoder_settings.encoder = &fake_encoder_; send_config_.encoder_settings.payload_name = "FAKE"; send_config_.encoder_settings.payload_type = kFakeSendPayloadType; @@ -97,7 +98,7 @@ void CallTest::CreateSendConfig(size_t num_streams) { void CallTest::CreateMatchingReceiveConfigs() { assert(!send_config_.rtp.ssrcs.empty()); assert(receive_configs_.empty()); - VideoReceiveStream::Config config = receiver_call_->GetDefaultReceiveConfig(); + VideoReceiveStream::Config config; VideoCodec codec = test::CreateDecoderVideoCodec(send_config_.encoder_settings); config.codecs.push_back(codec); @@ -121,7 +122,7 @@ void CallTest::CreateFrameGeneratorCapturer() { stream.width, stream.height, stream.max_framerate, - Clock::GetRealTimeClock())); + clock_)); } void CallTest::CreateStreams() { assert(send_stream_ == NULL); @@ -150,7 +151,8 @@ const unsigned int CallTest::kLongTimeoutMs = 120 * 1000; const uint8_t CallTest::kSendPayloadType = 100; const uint8_t CallTest::kFakeSendPayloadType = 125; const uint8_t CallTest::kSendRtxPayloadType = 98; -const uint32_t CallTest::kSendRtxSsrc = 0xBADCAFE; +const uint32_t CallTest::kSendRtxSsrcs[kNumSsrcs] = {0xBADCAFD, 0xBADCAFE, + 0xBADCAFF}; const uint32_t CallTest::kSendSsrcs[kNumSsrcs] = {0xC0FFED, 0xC0FFEE, 0xC0FFEF}; const uint32_t CallTest::kReceiverLocalSsrc = 0x123456; const int CallTest::kNackRtpHistoryMs = 1000; diff --git a/test/call_test.h b/test/call_test.h index 37a883d6..e24d8359 100644 --- a/test/call_test.h +++ b/test/call_test.h @@ -35,7 +35,7 @@ class CallTest : public ::testing::Test { static const uint8_t kSendPayloadType; static const uint8_t kSendRtxPayloadType; static const uint8_t kFakeSendPayloadType; - static const uint32_t kSendRtxSsrc; + static const uint32_t kSendRtxSsrcs[kNumSsrcs]; static const uint32_t kSendSsrcs[kNumSsrcs]; static const uint32_t kReceiverLocalSsrc; static const int kNackRtpHistoryMs; @@ -58,6 +58,8 @@ class CallTest : public ::testing::Test { void Stop(); void DestroyStreams(); + Clock* const clock_; + scoped_ptr<Call> sender_call_; VideoSendStream::Config send_config_; std::vector<VideoStream> video_streams_; diff --git a/test/encoder_settings.cc b/test/encoder_settings.cc index 5193be65..9842d1ed 100644 --- a/test/encoder_settings.cc +++ b/test/encoder_settings.cc @@ -59,9 +59,13 @@ VideoCodec CreateDecoderVideoCodec( codec.plType = encoder_settings.payload_type; strcpy(codec.plName, encoder_settings.payload_name.c_str()); - codec.codecType = - (encoder_settings.payload_name == "VP8" ? kVideoCodecVP8 - : kVideoCodecGeneric); + if (encoder_settings.payload_name == "VP8") { + codec.codecType = kVideoCodecVP8; + } else if (encoder_settings.payload_name == "H264") { + codec.codecType = kVideoCodecH264; + } else { + codec.codecType = kVideoCodecGeneric; + } if (codec.codecType == kVideoCodecVP8) { codec.codecSpecific.VP8.resilience = kResilientStream; @@ -73,6 +77,12 @@ VideoCodec CreateDecoderVideoCodec( codec.codecSpecific.VP8.keyFrameInterval = 3000; } + if (codec.codecType == kVideoCodecH264) { + codec.codecSpecific.H264.profile = kProfileBase; + codec.codecSpecific.H264.frameDroppingOn = true; + codec.codecSpecific.H264.keyFrameInterval = 3000; + } + codec.width = 320; codec.height = 180; codec.startBitrate = codec.minBitrate = codec.maxBitrate = 300; diff --git a/test/rtp_rtcp_observer.h b/test/rtp_rtcp_observer.h index 670a29de..11531b3b 100644 --- a/test/rtp_rtcp_observer.h +++ b/test/rtp_rtcp_observer.h @@ -129,7 +129,7 @@ class RtpRtcpObserver { private: virtual bool SendRtp(const uint8_t* packet, size_t length) OVERRIDE { - EXPECT_FALSE(RtpHeaderParser::IsRtcp(packet, static_cast<int>(length))); + EXPECT_FALSE(RtpHeaderParser::IsRtcp(packet, length)); Action action; { CriticalSectionScoped lock(crit_); @@ -146,7 +146,7 @@ class RtpRtcpObserver { } virtual bool SendRtcp(const uint8_t* packet, size_t length) OVERRIDE { - EXPECT_TRUE(RtpHeaderParser::IsRtcp(packet, static_cast<int>(length))); + EXPECT_TRUE(RtpHeaderParser::IsRtcp(packet, length)); Action action; { CriticalSectionScoped lock(crit_); diff --git a/video/OWNERS b/video/OWNERS index 78a39675..b5f9aeba 100644 --- a/video/OWNERS +++ b/video/OWNERS @@ -1,7 +1,6 @@ mflodman@webrtc.org stefan@webrtc.org -wu@webrtc.org -mallinath@webrtc.org +pbos@webrtc.org # These are for the common case of adding or renaming files. If you're doing # structural changes, please get a review from a reviewer in this file. diff --git a/video/bitrate_estimator_tests.cc b/video/bitrate_estimator_tests.cc index d21de179..1ec95c3b 100644 --- a/video/bitrate_estimator_tests.cc +++ b/video/bitrate_estimator_tests.cc @@ -27,6 +27,92 @@ #include "webrtc/test/frame_generator_capturer.h" namespace webrtc { +namespace { +// Note: consider to write tests that don't depend on the trace system instead +// of re-using this class. +class TraceObserver { + public: + TraceObserver() { + Trace::set_level_filter(kTraceTerseInfo); + + Trace::CreateTrace(); + Trace::SetTraceCallback(&callback_); + + // Call webrtc trace to initialize the tracer that would otherwise trigger a + // data-race if left to be initialized by multiple threads (i.e. threads + // spawned by test::DirectTransport members in BitrateEstimatorTest). + WEBRTC_TRACE(kTraceStateInfo, + kTraceUtility, + -1, + "Instantiate without data races."); + } + + ~TraceObserver() { + Trace::SetTraceCallback(NULL); + Trace::ReturnTrace(); + } + + void PushExpectedLogLine(const std::string& expected_log_line) { + callback_.PushExpectedLogLine(expected_log_line); + } + + EventTypeWrapper Wait() { + return callback_.Wait(); + } + + private: + class Callback : public TraceCallback { + public: + Callback() + : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), + done_(EventWrapper::Create()) {} + + virtual void Print(TraceLevel level, + const char* message, + int length) OVERRIDE { + CriticalSectionScoped lock(crit_sect_.get()); + std::string msg(message); + if (msg.find("BitrateEstimator") != std::string::npos) { + received_log_lines_.push_back(msg); + } + int num_popped = 0; + while (!received_log_lines_.empty() && !expected_log_lines_.empty()) { + std::string a = received_log_lines_.front(); + std::string b = expected_log_lines_.front(); + received_log_lines_.pop_front(); + expected_log_lines_.pop_front(); + num_popped++; + EXPECT_TRUE(a.find(b) != std::string::npos); + } + if (expected_log_lines_.size() <= 0) { + if (num_popped > 0) { + done_->Set(); + } + return; + } + } + + EventTypeWrapper Wait() { + return done_->Wait(test::CallTest::kDefaultTimeoutMs); + } + + void PushExpectedLogLine(const std::string& expected_log_line) { + CriticalSectionScoped lock(crit_sect_.get()); + expected_log_lines_.push_back(expected_log_line); + } + + private: + typedef std::list<std::string> Strings; + const scoped_ptr<CriticalSectionWrapper> crit_sect_; + Strings received_log_lines_ GUARDED_BY(crit_sect_); + Strings expected_log_lines_ GUARDED_BY(crit_sect_); + scoped_ptr<EventWrapper> done_; + + }; + + Callback callback_; +}; +} // namespace static const int kTOFExtensionId = 4; static const int kASTExtensionId = 5; @@ -48,11 +134,6 @@ class BitrateEstimatorTest : public test::CallTest { } virtual void SetUp() { - Trace::CreateTrace(); - Trace::SetTraceCallback(&receiver_trace_); - // Reduce the chance that spurious traces will ruin the test. - Trace::set_level_filter(kTraceTerseInfo); - Call::Config receiver_call_config(&receive_transport_); receiver_call_.reset(Call::Create(receiver_call_config)); @@ -62,7 +143,7 @@ class BitrateEstimatorTest : public test::CallTest { send_transport_.SetReceiver(receiver_call_->Receiver()); receive_transport_.SetReceiver(sender_call_->Receiver()); - send_config_ = sender_call_->GetDefaultSendConfig(); + send_config_ = VideoSendStream::Config(); send_config_.rtp.ssrcs.push_back(kSendSsrcs[0]); // Encoders will be set separately per stream. send_config_.encoder_settings.encoder = NULL; @@ -70,7 +151,7 @@ class BitrateEstimatorTest : public test::CallTest { send_config_.encoder_settings.payload_type = kFakeSendPayloadType; video_streams_ = test::CreateVideoStreams(1); - receive_config_ = receiver_call_->GetDefaultReceiveConfig(); + receive_config_ = VideoReceiveStream::Config(); assert(receive_config_.codecs.empty()); VideoCodec codec = test::CreateDecoderVideoCodec(send_config_.encoder_settings); @@ -97,63 +178,11 @@ class BitrateEstimatorTest : public test::CallTest { } receiver_call_.reset(); - - Trace::SetTraceCallback(NULL); - Trace::ReturnTrace(); } protected: friend class Stream; - class TraceObserver : public TraceCallback { - public: - TraceObserver() - : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), - received_log_lines_(), - expected_log_lines_(), - done_(EventWrapper::Create()) { - } - - void PushExpectedLogLine(const std::string& expected_log_line) { - CriticalSectionScoped lock(crit_sect_.get()); - expected_log_lines_.push_back(expected_log_line); - } - - virtual void Print(TraceLevel level, - const char* message, - int length) OVERRIDE { - CriticalSectionScoped lock(crit_sect_.get()); - std::string msg(message); - if (msg.find("BitrateEstimator") != std::string::npos) { - received_log_lines_.push_back(msg); - } - int num_popped = 0; - while (!received_log_lines_.empty() && !expected_log_lines_.empty()) { - std::string a = received_log_lines_.front(); - std::string b = expected_log_lines_.front(); - received_log_lines_.pop_front(); - expected_log_lines_.pop_front(); - num_popped++; - EXPECT_TRUE(a.find(b) != std::string::npos); - } - if (expected_log_lines_.size() <= 0) { - if (num_popped > 0) { - done_->Set(); - } - return; - } - } - - EventTypeWrapper Wait() { return done_->Wait(kDefaultTimeoutMs); } - - private: - typedef std::list<std::string> Strings; - const scoped_ptr<CriticalSectionWrapper> crit_sect_; - Strings received_log_lines_ GUARDED_BY(crit_sect_); - Strings expected_log_lines_ GUARDED_BY(crit_sect_); - scoped_ptr<EventWrapper> done_; - }; - class Stream { public: explicit Stream(BitrateEstimatorTest* test) diff --git a/video/call.cc b/video/call.cc index 6daa8b07..95e1c7b7 100644 --- a/video/call.cc +++ b/video/call.cc @@ -67,8 +67,6 @@ class Call : public webrtc::Call, public PacketReceiver { virtual PacketReceiver* Receiver() OVERRIDE; - virtual VideoSendStream::Config GetDefaultSendConfig() OVERRIDE; - virtual VideoSendStream* CreateVideoSendStream( const VideoSendStream::Config& config, const std::vector<VideoStream>& video_streams, @@ -77,8 +75,6 @@ class Call : public webrtc::Call, public PacketReceiver { virtual void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) OVERRIDE; - virtual VideoReceiveStream::Config GetDefaultReceiveConfig() OVERRIDE; - virtual VideoReceiveStream* CreateVideoReceiveStream( const VideoReceiveStream::Config& config) OVERRIDE; @@ -93,9 +89,7 @@ class Call : public webrtc::Call, public PacketReceiver { private: DeliveryStatus DeliverRtcp(const uint8_t* packet, size_t length); - DeliveryStatus DeliverRtp(const RTPHeader& header, - const uint8_t* packet, - size_t length); + DeliveryStatus DeliverRtp(const uint8_t* packet, size_t length); Call::Config config_; @@ -106,10 +100,10 @@ class Call : public webrtc::Call, public PacketReceiver { std::map<uint32_t, VideoSendStream*> send_ssrcs_ GUARDED_BY(send_lock_); scoped_ptr<RWLockWrapper> send_lock_; - scoped_ptr<RtpHeaderParser> rtp_header_parser_; - scoped_ptr<CpuOveruseObserverProxy> overuse_observer_proxy_; + VideoSendStream::RtpStateMap suspended_send_ssrcs_; + VideoEngine* video_engine_; ViERTP_RTCP* rtp_rtcp_; ViECodec* codec_; @@ -137,7 +131,6 @@ Call::Call(webrtc::VideoEngine* video_engine, const Call::Config& config) : config_(config), receive_lock_(RWLockWrapper::CreateRWLock()), send_lock_(RWLockWrapper::CreateRWLock()), - rtp_header_parser_(RtpHeaderParser::Create()), video_engine_(video_engine), base_channel_id_(-1) { assert(video_engine != NULL); @@ -173,11 +166,6 @@ Call::~Call() { PacketReceiver* Call::Receiver() { return this; } -VideoSendStream::Config Call::GetDefaultSendConfig() { - VideoSendStream::Config config; - return config; -} - VideoSendStream* Call::CreateVideoSendStream( const VideoSendStream::Config& config, const std::vector<VideoStream>& video_streams, @@ -193,6 +181,7 @@ VideoSendStream* Call::CreateVideoSendStream( config, video_streams, encoder_settings, + suspended_send_ssrcs_, base_channel_id_, config_.start_bitrate_bps != -1 ? config_.start_bitrate_bps : kDefaultVideoStreamBitrateBps); @@ -208,31 +197,34 @@ VideoSendStream* Call::CreateVideoSendStream( void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) { assert(send_stream != NULL); + send_stream->Stop(); + VideoSendStream* send_stream_impl = NULL; { WriteLockScoped write_lock(*send_lock_); - for (std::map<uint32_t, VideoSendStream*>::iterator it = - send_ssrcs_.begin(); - it != send_ssrcs_.end(); - ++it) { + std::map<uint32_t, VideoSendStream*>::iterator it = send_ssrcs_.begin(); + while (it != send_ssrcs_.end()) { if (it->second == static_cast<VideoSendStream*>(send_stream)) { send_stream_impl = it->second; - send_ssrcs_.erase(it); - break; + send_ssrcs_.erase(it++); + } else { + ++it; } } } + VideoSendStream::RtpStateMap rtp_state = send_stream_impl->GetRtpStates(); + + for (VideoSendStream::RtpStateMap::iterator it = rtp_state.begin(); + it != rtp_state.end(); + ++it) { + suspended_send_ssrcs_[it->first] = it->second; + } + assert(send_stream_impl != NULL); delete send_stream_impl; } -VideoReceiveStream::Config Call::GetDefaultReceiveConfig() { - VideoReceiveStream::Config config; - config.rtp.remb = true; - return config; -} - VideoReceiveStream* Call::CreateVideoReceiveStream( const VideoReceiveStream::Config& config) { VideoReceiveStream* receive_stream = @@ -291,7 +283,7 @@ uint32_t Call::ReceiveBitrateEstimate() { return 0; } -Call::PacketReceiver::DeliveryStatus Call::DeliverRtcp(const uint8_t* packet, +PacketReceiver::DeliveryStatus Call::DeliverRtcp(const uint8_t* packet, size_t length) { // TODO(pbos): Figure out what channel needs it actually. // Do NOT broadcast! Also make sure it's a valid packet. @@ -322,32 +314,32 @@ Call::PacketReceiver::DeliveryStatus Call::DeliverRtcp(const uint8_t* packet, return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR; } -Call::PacketReceiver::DeliveryStatus Call::DeliverRtp(const RTPHeader& header, - const uint8_t* packet, - size_t length) { +PacketReceiver::DeliveryStatus Call::DeliverRtp(const uint8_t* packet, + size_t length) { + // Minimum RTP header size. + if (length < 12) + return DELIVERY_PACKET_ERROR; + + const uint8_t* ptr = &packet[8]; + uint32_t ssrc = ptr[0] << 24 | ptr[1] << 16 | ptr[2] << 8 | ptr[3] ; + ReadLockScoped read_lock(*receive_lock_); std::map<uint32_t, VideoReceiveStream*>::iterator it = - receive_ssrcs_.find(header.ssrc); + receive_ssrcs_.find(ssrc); if (it == receive_ssrcs_.end()) return DELIVERY_UNKNOWN_SSRC; - return it->second->DeliverRtp(static_cast<const uint8_t*>(packet), length) - ? DELIVERY_OK - : DELIVERY_PACKET_ERROR; + return it->second->DeliverRtp(packet, length) ? DELIVERY_OK + : DELIVERY_PACKET_ERROR; } -Call::PacketReceiver::DeliveryStatus Call::DeliverPacket(const uint8_t* packet, - size_t length) { - // TODO(pbos): ExtensionMap if there are extensions. - if (RtpHeaderParser::IsRtcp(packet, static_cast<int>(length))) +PacketReceiver::DeliveryStatus Call::DeliverPacket(const uint8_t* packet, + size_t length) { + if (RtpHeaderParser::IsRtcp(packet, length)) return DeliverRtcp(packet, length); - RTPHeader rtp_header; - if (!rtp_header_parser_->Parse(packet, static_cast<int>(length), &rtp_header)) - return DELIVERY_PACKET_ERROR; - - return DeliverRtp(rtp_header, packet, length); + return DeliverRtp(packet, length); } } // namespace internal diff --git a/video/call_perf_tests.cc b/video/call_perf_tests.cc index 18246168..62d2adcb 100644 --- a/video/call_perf_tests.cc +++ b/video/call_perf_tests.cc @@ -354,7 +354,7 @@ void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config, virtual Action OnSendRtp(const uint8_t* packet, size_t length) { RTPHeader header; - EXPECT_TRUE(parser_->Parse(packet, static_cast<int>(length), &header)); + EXPECT_TRUE(parser_->Parse(packet, length, &header)); if (!rtp_start_timestamp_set_) { // Calculate the rtp timestamp offset in order to calculate the real diff --git a/video/end_to_end_tests.cc b/video/end_to_end_tests.cc index fa3d3cf4..8946f890 100644 --- a/video/end_to_end_tests.cc +++ b/video/end_to_end_tests.cc @@ -58,6 +58,7 @@ class EndToEndTest : public test::CallTest { void RespectsRtcpMode(newapi::RtcpMode rtcp_mode); void TestXrReceiverReferenceTimeReport(bool enable_rrtr); void TestSendsSetSsrcs(size_t num_ssrcs, bool send_single_ssrc_first); + void TestRtpStatePreservation(bool use_rtx); }; TEST_F(EndToEndTest, ReceiverCanBeStartedTwice) { @@ -253,8 +254,7 @@ TEST_F(EndToEndTest, ReceivesAndRetransmitsNack) { private: virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { RTPHeader header; - EXPECT_TRUE( - rtp_parser_->Parse(packet, static_cast<int>(length), &header)); + EXPECT_TRUE(rtp_parser_->Parse(packet, length, &header)); // Never drop retransmitted packets. if (dropped_packets_.find(header.sequenceNumber) != @@ -341,7 +341,7 @@ TEST_F(EndToEndTest, DISABLED_CanReceiveFec) { virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE EXCLUSIVE_LOCKS_REQUIRED(crit_) { RTPHeader header; - EXPECT_TRUE(parser_->Parse(packet, static_cast<int>(length), &header)); + EXPECT_TRUE(parser_->Parse(packet, length, &header)); EXPECT_EQ(kRedPayloadType, header.payloadType); int encapsulated_payload_type = @@ -434,7 +434,7 @@ void EndToEndTest::DecodesRetransmittedFrame(bool retransmit_over_rtx) { public: explicit RetransmissionObserver(bool expect_rtx) : EndToEndTest(kDefaultTimeoutMs), - retransmission_ssrc_(expect_rtx ? kSendRtxSsrc : kSendSsrcs[0]), + retransmission_ssrc_(expect_rtx ? kSendRtxSsrcs[0] : kSendSsrcs[0]), retransmission_payload_type_(expect_rtx ? kSendRtxPayloadType : kFakeSendPayloadType), marker_bits_observed_(0), @@ -444,7 +444,7 @@ void EndToEndTest::DecodesRetransmittedFrame(bool retransmit_over_rtx) { private: virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { RTPHeader header; - EXPECT_TRUE(parser_->Parse(packet, static_cast<int>(length), &header)); + EXPECT_TRUE(parser_->Parse(packet, length, &header)); if (header.timestamp == retransmitted_timestamp_) { EXPECT_EQ(retransmission_ssrc_, header.ssrc); @@ -481,10 +481,11 @@ void EndToEndTest::DecodesRetransmittedFrame(bool retransmit_over_rtx) { send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; (*receive_configs)[0].pre_render_callback = this; (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; - if (retransmission_ssrc_ == kSendRtxSsrc) { - send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrc); + if (retransmission_ssrc_ == kSendRtxSsrcs[0]) { + send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]); send_config->rtp.rtx.payload_type = kSendRtxPayloadType; - (*receive_configs)[0].rtp.rtx[kSendRtxPayloadType].ssrc = kSendRtxSsrc; + (*receive_configs)[0].rtp.rtx[kSendRtxPayloadType].ssrc = + kSendRtxSsrcs[0]; (*receive_configs)[0].rtp.rtx[kSendRtxPayloadType].payload_type = kSendRtxPayloadType; } @@ -630,7 +631,7 @@ void EndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) { private: virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { RTPHeader header; - EXPECT_TRUE(parser_->Parse(packet, static_cast<int>(length), &header)); + EXPECT_TRUE(parser_->Parse(packet, length, &header)); // Drop all retransmitted packets to force a PLI. if (header.timestamp <= highest_dropped_timestamp_) @@ -722,7 +723,7 @@ TEST_F(EndToEndTest, UnknownRtpPacketGivesUnknownSsrcReturnCode) { private: virtual DeliveryStatus DeliverPacket(const uint8_t* packet, size_t length) OVERRIDE { - if (RtpHeaderParser::IsRtcp(packet, static_cast<int>(length))) { + if (RtpHeaderParser::IsRtcp(packet, length)) { return receiver_->DeliverPacket(packet, length); } else { DeliveryStatus delivery_status = @@ -917,7 +918,7 @@ TEST_F(EndToEndTest, SendsAndReceivesMultipleStreams) { int height = codec_settings[i].height; observers[i] = new VideoOutputObserver(&frame_generators[i], width, height); - VideoSendStream::Config send_config = sender_call->GetDefaultSendConfig(); + VideoSendStream::Config send_config; send_config.rtp.ssrcs.push_back(ssrc); send_config.encoder_settings.encoder = encoders[i].get(); send_config.encoder_settings.payload_name = "VP8"; @@ -933,8 +934,7 @@ TEST_F(EndToEndTest, SendsAndReceivesMultipleStreams) { sender_call->CreateVideoSendStream(send_config, video_streams, NULL); send_streams[i]->Start(); - VideoReceiveStream::Config receive_config = - receiver_call->GetDefaultReceiveConfig(); + VideoReceiveStream::Config receive_config; receive_config.renderer = observers[i]; receive_config.rtp.remote_ssrc = ssrc; receive_config.rtp.local_ssrc = kReceiverLocalSsrc; @@ -1187,7 +1187,7 @@ void EndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs, private: virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { RTPHeader header; - EXPECT_TRUE(parser_->Parse(packet, static_cast<int>(length), &header)); + EXPECT_TRUE(parser_->Parse(packet, length, &header)); EXPECT_TRUE(valid_ssrcs_[header.ssrc]) << "Received unknown SSRC: " << header.ssrc; @@ -1543,4 +1543,251 @@ TEST_F(EndToEndTest, CanSwitchToUseAllSsrcs) { TestSendsSetSsrcs(kNumSsrcs, true); } +TEST_F(EndToEndTest, RedundantPayloadsTransmittedOnAllSsrcs) { + class ObserveRedundantPayloads: public test::EndToEndTest { + public: + ObserveRedundantPayloads() + : EndToEndTest(kDefaultTimeoutMs), ssrcs_to_observe_(kNumSsrcs) { + for(size_t i = 0; i < kNumSsrcs; ++i) { + registered_rtx_ssrc_[kSendRtxSsrcs[i]] = true; + } + } + + private: + virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { + RTPHeader header; + EXPECT_TRUE(parser_->Parse(packet, length, &header)); + + if (!registered_rtx_ssrc_[header.ssrc]) + return SEND_PACKET; + + EXPECT_LE(static_cast<size_t>(header.headerLength + header.paddingLength), + length); + const bool packet_is_redundant_payload = + static_cast<size_t>(header.headerLength + header.paddingLength) < + length; + + if (!packet_is_redundant_payload) + return SEND_PACKET; + + if (!observed_redundant_retransmission_[header.ssrc]) { + observed_redundant_retransmission_[header.ssrc] = true; + if (--ssrcs_to_observe_ == 0) + observation_complete_->Set(); + } + + return SEND_PACKET; + } + + virtual size_t GetNumStreams() const OVERRIDE { return kNumSsrcs; } + + virtual void ModifyConfigs( + VideoSendStream::Config* send_config, + std::vector<VideoReceiveStream::Config>* receive_configs, + std::vector<VideoStream>* video_streams) OVERRIDE { + // Set low simulcast bitrates to not have to wait for bandwidth ramp-up. + for (size_t i = 0; i < video_streams->size(); ++i) { + (*video_streams)[i].min_bitrate_bps = 10000; + (*video_streams)[i].target_bitrate_bps = 15000; + (*video_streams)[i].max_bitrate_bps = 20000; + } + // Significantly higher than max bitrates for all video streams -> forcing + // padding to trigger redundant padding on all RTX SSRCs. + send_config->rtp.min_transmit_bitrate_bps = 100000; + + send_config->rtp.rtx.payload_type = kSendRtxPayloadType; + send_config->rtp.rtx.pad_with_redundant_payloads = true; + + for (size_t i = 0; i < kNumSsrcs; ++i) + send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]); + } + + virtual void PerformTest() OVERRIDE { + EXPECT_EQ(kEventSignaled, Wait()) + << "Timed out while waiting for redundant payloads on all SSRCs."; + } + + private: + size_t ssrcs_to_observe_; + std::map<uint32_t, bool> observed_redundant_retransmission_; + std::map<uint32_t, bool> registered_rtx_ssrc_; + } test; + + RunBaseTest(&test); +} + +void EndToEndTest::TestRtpStatePreservation(bool use_rtx) { + static const uint32_t kMaxSequenceNumberGap = 100; + static const uint64_t kMaxTimestampGap = kDefaultTimeoutMs * 90; + class RtpSequenceObserver : public test::RtpRtcpObserver { + public: + RtpSequenceObserver(bool use_rtx) + : test::RtpRtcpObserver(kDefaultTimeoutMs), + crit_(CriticalSectionWrapper::CreateCriticalSection()), + ssrcs_to_observe_(kNumSsrcs) { + for (size_t i = 0; i < kNumSsrcs; ++i) { + configured_ssrcs_[kSendSsrcs[i]] = true; + if (use_rtx) + configured_ssrcs_[kSendRtxSsrcs[i]] = true; + } + } + + void ResetExpectedSsrcs(size_t num_expected_ssrcs) { + CriticalSectionScoped lock(crit_.get()); + ssrc_observed_.clear(); + ssrcs_to_observe_ = num_expected_ssrcs; + } + + private: + virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { + RTPHeader header; + EXPECT_TRUE(parser_->Parse(packet, length, &header)); + const uint32_t ssrc = header.ssrc; + const uint16_t sequence_number = header.sequenceNumber; + const uint32_t timestamp = header.timestamp; + const bool only_padding = + static_cast<size_t>(header.headerLength + header.paddingLength) == + length; + + EXPECT_TRUE(configured_ssrcs_[ssrc]) + << "Received SSRC that wasn't configured: " << ssrc; + + std::map<uint32_t, uint16_t>::iterator it = + last_observed_sequence_number_.find(header.ssrc); + if (it == last_observed_sequence_number_.end()) { + last_observed_sequence_number_[ssrc] = sequence_number; + last_observed_timestamp_[ssrc] = timestamp; + } else { + // Verify sequence numbers are reasonably close. + uint32_t extended_sequence_number = sequence_number; + // Check for roll-over. + if (sequence_number < last_observed_sequence_number_[ssrc]) + extended_sequence_number += 0xFFFFu + 1; + EXPECT_LE( + extended_sequence_number - last_observed_sequence_number_[ssrc], + kMaxSequenceNumberGap) + << "Gap in sequence numbers (" + << last_observed_sequence_number_[ssrc] << " -> " << sequence_number + << ") too large for SSRC: " << ssrc << "."; + last_observed_sequence_number_[ssrc] = sequence_number; + + // TODO(pbos): Remove this check if we ever have monotonically + // increasing timestamps. Right now padding packets add a delta which + // can cause reordering between padding packets and regular packets, + // hence we drop padding-only packets to not flake. + if (only_padding) { + // Verify that timestamps are reasonably close. + uint64_t extended_timestamp = timestamp; + // Check for roll-over. + if (timestamp < last_observed_timestamp_[ssrc]) + extended_timestamp += static_cast<uint64_t>(0xFFFFFFFFu) + 1; + EXPECT_LE(extended_timestamp - last_observed_timestamp_[ssrc], + kMaxTimestampGap) + << "Gap in timestamps (" << last_observed_timestamp_[ssrc] + << " -> " << timestamp << ") too large for SSRC: " << ssrc << "."; + } + last_observed_timestamp_[ssrc] = timestamp; + } + + CriticalSectionScoped lock(crit_.get()); + // Wait for media packets on all ssrcs. + if (!ssrc_observed_[ssrc] && !only_padding) { + ssrc_observed_[ssrc] = true; + if (--ssrcs_to_observe_ == 0) + observation_complete_->Set(); + } + + return SEND_PACKET; + } + + std::map<uint32_t, uint16_t> last_observed_sequence_number_; + std::map<uint32_t, uint32_t> last_observed_timestamp_; + std::map<uint32_t, bool> configured_ssrcs_; + + scoped_ptr<CriticalSectionWrapper> crit_; + size_t ssrcs_to_observe_ GUARDED_BY(crit_); + std::map<uint32_t, bool> ssrc_observed_ GUARDED_BY(crit_); + } observer(use_rtx); + + CreateCalls(Call::Config(observer.SendTransport()), + Call::Config(observer.ReceiveTransport())); + observer.SetReceivers(sender_call_->Receiver(), NULL); + + CreateSendConfig(kNumSsrcs); + + if (use_rtx) { + for (size_t i = 0; i < kNumSsrcs; ++i) { + send_config_.rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]); + } + send_config_.rtp.rtx.payload_type = kSendRtxPayloadType; + } + + // Lower bitrates so that all streams send initially. + for (size_t i = 0; i < video_streams_.size(); ++i) { + video_streams_[i].min_bitrate_bps = 10000; + video_streams_[i].target_bitrate_bps = 15000; + video_streams_[i].max_bitrate_bps = 20000; + } + + CreateMatchingReceiveConfigs(); + + CreateStreams(); + CreateFrameGeneratorCapturer(); + + Start(); + EXPECT_EQ(kEventSignaled, observer.Wait()) + << "Timed out waiting for all SSRCs to send packets."; + + // Test stream resetting more than once to make sure that the state doesn't + // get set once (this could be due to using std::map::insert for instance). + for (size_t i = 0; i < 3; ++i) { + frame_generator_capturer_->Stop(); + sender_call_->DestroyVideoSendStream(send_stream_); + + // Re-create VideoSendStream with only one stream. + std::vector<VideoStream> one_stream = video_streams_; + one_stream.resize(1); + send_stream_ = + sender_call_->CreateVideoSendStream(send_config_, one_stream, NULL); + send_stream_->Start(); + CreateFrameGeneratorCapturer(); + frame_generator_capturer_->Start(); + + observer.ResetExpectedSsrcs(1); + EXPECT_EQ(kEventSignaled, observer.Wait()) + << "Timed out waiting for single RTP packet."; + + // Reconfigure back to use all streams. + send_stream_->ReconfigureVideoEncoder(video_streams_, NULL); + observer.ResetExpectedSsrcs(kNumSsrcs); + EXPECT_EQ(kEventSignaled, observer.Wait()) + << "Timed out waiting for all SSRCs to send packets."; + + // Reconfigure down to one stream. + send_stream_->ReconfigureVideoEncoder(one_stream, NULL); + observer.ResetExpectedSsrcs(1); + EXPECT_EQ(kEventSignaled, observer.Wait()) + << "Timed out waiting for single RTP packet."; + + // Reconfigure back to use all streams. + send_stream_->ReconfigureVideoEncoder(video_streams_, NULL); + observer.ResetExpectedSsrcs(kNumSsrcs); + EXPECT_EQ(kEventSignaled, observer.Wait()) + << "Timed out waiting for all SSRCs to send packets."; + } + + observer.StopSending(); + + Stop(); + DestroyStreams(); +} + +TEST_F(EndToEndTest, RestartingSendStreamPreservesRtpState) { + TestRtpStatePreservation(false); +} + +TEST_F(EndToEndTest, RestartingSendStreamPreservesRtpStatesWithRtx) { + TestRtpStatePreservation(true); +} + } // namespace webrtc diff --git a/video/full_stack.cc b/video/full_stack.cc index e2a8e863..284efe20 100644 --- a/video/full_stack.cc +++ b/video/full_stack.cc @@ -104,7 +104,7 @@ class VideoAnalyzer : public PacketReceiver, size_t length) OVERRIDE { scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create()); RTPHeader header; - parser->Parse(packet, static_cast<int>(length), &header); + parser->Parse(packet, length, &header); { CriticalSectionScoped lock(crit_.get()); recv_times_[header.timestamp - rtp_timestamp_delta_] = @@ -143,7 +143,7 @@ class VideoAnalyzer : public PacketReceiver, virtual bool SendRtp(const uint8_t* packet, size_t length) OVERRIDE { scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create()); RTPHeader header; - parser->Parse(packet, static_cast<int>(length), &header); + parser->Parse(packet, length, &header); { CriticalSectionScoped lock(crit_.get()); diff --git a/video/loopback.cc b/video/loopback.cc index ea65ebb4..614ef47a 100644 --- a/video/loopback.cc +++ b/video/loopback.cc @@ -48,6 +48,9 @@ size_t StartBitrate() { return static_cast<size_t>(FLAGS_start_bitrate); } DEFINE_int32(max_bitrate, 800, "Maximum video bitrate."); size_t MaxBitrate() { return static_cast<size_t>(FLAGS_max_bitrate); } + +DEFINE_string(codec, "VP8", "Video codec to use."); +std::string Codec() { return static_cast<std::string>(FLAGS_codec); } } // namespace flags static const uint32_t kSendSsrc = 0x654321; @@ -68,14 +71,20 @@ void Loopback() { // Loopback, call sends to itself. transport.SetReceiver(call->Receiver()); - VideoSendStream::Config send_config = call->GetDefaultSendConfig(); + VideoSendStream::Config send_config; send_config.rtp.ssrcs.push_back(kSendSsrc); send_config.local_renderer = local_preview.get(); - - scoped_ptr<VP8Encoder> encoder(VP8Encoder::Create()); + scoped_ptr<VideoEncoder> encoder; + if (flags::Codec() == "VP8") { + encoder.reset(VP8Encoder::Create()); + } else { + // Codec not supported. + assert(false && "Codec not supported!"); + return; + } send_config.encoder_settings.encoder = encoder.get(); - send_config.encoder_settings.payload_name = "VP8"; + send_config.encoder_settings.payload_name = flags::Codec(); send_config.encoder_settings.payload_type = 124; std::vector<VideoStream> video_streams = test::CreateVideoStreams(1); VideoStream* stream = &video_streams[0]; @@ -99,7 +108,7 @@ void Loopback() { flags::Fps(), test_clock)); - VideoReceiveStream::Config receive_config = call->GetDefaultReceiveConfig(); + VideoReceiveStream::Config receive_config; receive_config.rtp.remote_ssrc = send_config.rtp.ssrcs[0]; receive_config.rtp.local_ssrc = kReceiverLocalSsrc; receive_config.renderer = loopback_video.get(); diff --git a/video/rampup_tests.cc b/video/rampup_tests.cc index 5529f92e..b6f63814 100644 --- a/video/rampup_tests.cc +++ b/video/rampup_tests.cc @@ -7,549 +7,496 @@ * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ -#include <assert.h> - -#include <map> -#include <string> -#include <vector> #include "testing/gtest/include/gtest/gtest.h" - -#include "webrtc/call.h" -#include "webrtc/common.h" -#include "webrtc/experiments.h" -#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" #include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" -#include "webrtc/system_wrappers/interface/event_wrapper.h" -#include "webrtc/system_wrappers/interface/scoped_ptr.h" -#include "webrtc/test/call_test.h" -#include "webrtc/test/direct_transport.h" -#include "webrtc/test/encoder_settings.h" -#include "webrtc/test/fake_decoder.h" -#include "webrtc/test/fake_encoder.h" -#include "webrtc/test/frame_generator_capturer.h" #include "webrtc/test/testsupport/perf_test.h" -#include "webrtc/video/transport_adapter.h" +#include "webrtc/video/rampup_tests.h" namespace webrtc { - namespace { -static const int kTransmissionTimeOffsetExtensionId = 6; + static const int kMaxPacketSize = 1500; -static const unsigned int kSingleStreamTargetBps = 1000000; - -class StreamObserver : public newapi::Transport, public RemoteBitrateObserver { - public: - typedef std::map<uint32_t, int> BytesSentMap; - typedef std::map<uint32_t, uint32_t> SsrcMap; - StreamObserver(const SsrcMap& rtx_media_ssrcs, - newapi::Transport* feedback_transport, - Clock* clock) - : clock_(clock), - test_done_(EventWrapper::Create()), - rtp_parser_(RtpHeaderParser::Create()), - feedback_transport_(feedback_transport), - receive_stats_(ReceiveStatistics::Create(clock)), - payload_registry_( - new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(false))), - crit_(CriticalSectionWrapper::CreateCriticalSection()), - expected_bitrate_bps_(0), - start_bitrate_bps_(0), - rtx_media_ssrcs_(rtx_media_ssrcs), - total_sent_(0), - padding_sent_(0), - rtx_media_sent_(0), - total_packets_sent_(0), - padding_packets_sent_(0), - rtx_media_packets_sent_(0) { - // Ideally we would only have to instantiate an RtcpSender, an - // RtpHeaderParser and a RemoteBitrateEstimator here, but due to the current - // state of the RTP module we need a full module and receive statistics to - // be able to produce an RTCP with REMB. - RtpRtcp::Configuration config; - config.receive_statistics = receive_stats_.get(); - feedback_transport_.Enable(); - config.outgoing_transport = &feedback_transport_; - rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config)); - rtp_rtcp_->SetREMBStatus(true); - rtp_rtcp_->SetRTCPStatus(kRtcpNonCompound); - rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset, - kTransmissionTimeOffsetExtensionId); - AbsoluteSendTimeRemoteBitrateEstimatorFactory rbe_factory; - const uint32_t kRemoteBitrateEstimatorMinBitrateBps = 30000; - remote_bitrate_estimator_.reset( - rbe_factory.Create(this, clock, kMimdControl, - kRemoteBitrateEstimatorMinBitrateBps)); - } - void set_expected_bitrate_bps(unsigned int expected_bitrate_bps) { - CriticalSectionScoped lock(crit_.get()); - expected_bitrate_bps_ = expected_bitrate_bps; - } +std::vector<uint32_t> GenerateSsrcs(size_t num_streams, + uint32_t ssrc_offset) { + std::vector<uint32_t> ssrcs; + for (size_t i = 0; i != num_streams; ++i) + ssrcs.push_back(static_cast<uint32_t>(ssrc_offset + i)); + return ssrcs; +} +} // namespace - void set_start_bitrate_bps(unsigned int start_bitrate_bps) { - CriticalSectionScoped lock(crit_.get()); - start_bitrate_bps_ = start_bitrate_bps; - } +StreamObserver::StreamObserver(const SsrcMap& rtx_media_ssrcs, + newapi::Transport* feedback_transport, + Clock* clock, + RemoteBitrateEstimatorFactory* rbe_factory, + RateControlType control_type) + : clock_(clock), + test_done_(EventWrapper::Create()), + rtp_parser_(RtpHeaderParser::Create()), + feedback_transport_(feedback_transport), + receive_stats_(ReceiveStatistics::Create(clock)), + payload_registry_( + new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(false))), + crit_(CriticalSectionWrapper::CreateCriticalSection()), + expected_bitrate_bps_(0), + start_bitrate_bps_(0), + rtx_media_ssrcs_(rtx_media_ssrcs), + total_sent_(0), + padding_sent_(0), + rtx_media_sent_(0), + total_packets_sent_(0), + padding_packets_sent_(0), + rtx_media_packets_sent_(0), + test_start_ms_(clock_->TimeInMilliseconds()), + ramp_up_finished_ms_(0) { + // Ideally we would only have to instantiate an RtcpSender, an + // RtpHeaderParser and a RemoteBitrateEstimator here, but due to the current + // state of the RTP module we need a full module and receive statistics to + // be able to produce an RTCP with REMB. + RtpRtcp::Configuration config; + config.receive_statistics = receive_stats_.get(); + feedback_transport_.Enable(); + config.outgoing_transport = &feedback_transport_; + rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config)); + rtp_rtcp_->SetREMBStatus(true); + rtp_rtcp_->SetRTCPStatus(kRtcpNonCompound); + rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime, + kAbsSendTimeExtensionId); + rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset, + kTransmissionTimeOffsetExtensionId); + const uint32_t kRemoteBitrateEstimatorMinBitrateBps = 30000; + remote_bitrate_estimator_.reset( + rbe_factory->Create(this, clock, control_type, + kRemoteBitrateEstimatorMinBitrateBps)); +} - virtual void OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs, - unsigned int bitrate) OVERRIDE { - CriticalSectionScoped lock(crit_.get()); - assert(expected_bitrate_bps_ > 0); - if (start_bitrate_bps_ != 0) { - // For tests with an explicitly set start bitrate, verify the first - // bitrate estimate is close to the start bitrate and lower than the - // test target bitrate. This is to verify a call respects the configured - // start bitrate, but due to the BWE implementation we can't guarantee the - // first estimate really is as high as the start bitrate. - EXPECT_GT(bitrate, 0.9 * start_bitrate_bps_); - EXPECT_LT(bitrate, expected_bitrate_bps_); - start_bitrate_bps_ = 0; - } - if (bitrate >= expected_bitrate_bps_) { - // Just trigger if there was any rtx padding packet. - if (rtx_media_ssrcs_.empty() || rtx_media_sent_ > 0) { - TriggerTestDone(); - } - } - rtp_rtcp_->SetREMBData( - bitrate, static_cast<uint8_t>(ssrcs.size()), &ssrcs[0]); - rtp_rtcp_->Process(); - } +void StreamObserver::set_expected_bitrate_bps( + unsigned int expected_bitrate_bps) { + CriticalSectionScoped lock(crit_.get()); + expected_bitrate_bps_ = expected_bitrate_bps; +} - virtual bool SendRtp(const uint8_t* packet, size_t length) OVERRIDE { - CriticalSectionScoped lock(crit_.get()); - RTPHeader header; - EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length), &header)); - receive_stats_->IncomingPacket(header, length, false); - payload_registry_->SetIncomingPayloadType(header); - remote_bitrate_estimator_->IncomingPacket( - clock_->TimeInMilliseconds(), static_cast<int>(length - 12), header); - if (remote_bitrate_estimator_->TimeUntilNextProcess() <= 0) { - remote_bitrate_estimator_->Process(); - } - total_sent_ += length; - padding_sent_ += header.paddingLength; - ++total_packets_sent_; - if (header.paddingLength > 0) - ++padding_packets_sent_; - if (rtx_media_ssrcs_.find(header.ssrc) != rtx_media_ssrcs_.end()) { - rtx_media_sent_ += length - header.headerLength - header.paddingLength; - if (header.paddingLength == 0) - ++rtx_media_packets_sent_; - uint8_t restored_packet[kMaxPacketSize]; - uint8_t* restored_packet_ptr = restored_packet; - int restored_length = static_cast<int>(length); - payload_registry_->RestoreOriginalPacket(&restored_packet_ptr, - packet, - &restored_length, - rtx_media_ssrcs_[header.ssrc], - header); - length = restored_length; - EXPECT_TRUE(rtp_parser_->Parse( - restored_packet, static_cast<int>(length), &header)); - } else { - rtp_rtcp_->SetRemoteSSRC(header.ssrc); +void StreamObserver::set_start_bitrate_bps(unsigned int start_bitrate_bps) { + CriticalSectionScoped lock(crit_.get()); + start_bitrate_bps_ = start_bitrate_bps; +} + +void StreamObserver::OnReceiveBitrateChanged( + const std::vector<unsigned int>& ssrcs, unsigned int bitrate) { + CriticalSectionScoped lock(crit_.get()); + assert(expected_bitrate_bps_ > 0); + if (start_bitrate_bps_ != 0) { + // For tests with an explicitly set start bitrate, verify the first + // bitrate estimate is close to the start bitrate and lower than the + // test target bitrate. This is to verify a call respects the configured + // start bitrate, but due to the BWE implementation we can't guarantee the + // first estimate really is as high as the start bitrate. + EXPECT_GT(bitrate, 0.9 * start_bitrate_bps_); + EXPECT_LT(bitrate, expected_bitrate_bps_); + start_bitrate_bps_ = 0; + } + if (bitrate >= expected_bitrate_bps_) { + ramp_up_finished_ms_ = clock_->TimeInMilliseconds(); + // Just trigger if there was any rtx padding packet. + if (rtx_media_ssrcs_.empty() || rtx_media_sent_ > 0) { + TriggerTestDone(); } - return true; } + rtp_rtcp_->SetREMBData( + bitrate, static_cast<uint8_t>(ssrcs.size()), &ssrcs[0]); + rtp_rtcp_->Process(); +} - virtual bool SendRtcp(const uint8_t* packet, size_t length) OVERRIDE { - return true; +bool StreamObserver::SendRtp(const uint8_t* packet, size_t length) { + CriticalSectionScoped lock(crit_.get()); + RTPHeader header; + EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length), &header)); + receive_stats_->IncomingPacket(header, length, false); + payload_registry_->SetIncomingPayloadType(header); + remote_bitrate_estimator_->IncomingPacket( + clock_->TimeInMilliseconds(), static_cast<int>(length - 12), header); + if (remote_bitrate_estimator_->TimeUntilNextProcess() <= 0) { + remote_bitrate_estimator_->Process(); + } + total_sent_ += length; + padding_sent_ += header.paddingLength; + ++total_packets_sent_; + if (header.paddingLength > 0) + ++padding_packets_sent_; + if (rtx_media_ssrcs_.find(header.ssrc) != rtx_media_ssrcs_.end()) { + rtx_media_sent_ += length - header.headerLength - header.paddingLength; + if (header.paddingLength == 0) + ++rtx_media_packets_sent_; + uint8_t restored_packet[kMaxPacketSize]; + uint8_t* restored_packet_ptr = restored_packet; + int restored_length = static_cast<int>(length); + payload_registry_->RestoreOriginalPacket(&restored_packet_ptr, + packet, + &restored_length, + rtx_media_ssrcs_[header.ssrc], + header); + length = restored_length; + EXPECT_TRUE(rtp_parser_->Parse( + restored_packet, static_cast<int>(length), &header)); + } else { + rtp_rtcp_->SetRemoteSSRC(header.ssrc); } + return true; +} - EventTypeWrapper Wait() { return test_done_->Wait(120 * 1000); } +bool StreamObserver::SendRtcp(const uint8_t* packet, size_t length) { + return true; +} - private: - void ReportResult(const std::string& measurement, - size_t value, - const std::string& units) { - webrtc::test::PrintResult( - measurement, "", - ::testing::UnitTest::GetInstance()->current_test_info()->name(), - value, units, false); - } +EventTypeWrapper StreamObserver::Wait() { return test_done_->Wait(120 * 1000); } - void TriggerTestDone() EXCLUSIVE_LOCKS_REQUIRED(crit_) { - ReportResult("total-sent", total_sent_, "bytes"); - ReportResult("padding-sent", padding_sent_, "bytes"); - ReportResult("rtx-media-sent", rtx_media_sent_, "bytes"); - ReportResult("total-packets-sent", total_packets_sent_, "packets"); - ReportResult("padding-packets-sent", padding_packets_sent_, "packets"); - ReportResult("rtx-packets-sent", rtx_media_packets_sent_, "packets"); - test_done_->Set(); - } +void StreamObserver::ReportResult(const std::string& measurement, + size_t value, + const std::string& units) { + webrtc::test::PrintResult( + measurement, "", + ::testing::UnitTest::GetInstance()->current_test_info()->name(), + value, units, false); +} - Clock* const clock_; - const scoped_ptr<EventWrapper> test_done_; - const scoped_ptr<RtpHeaderParser> rtp_parser_; - scoped_ptr<RtpRtcp> rtp_rtcp_; - internal::TransportAdapter feedback_transport_; - const scoped_ptr<ReceiveStatistics> receive_stats_; - const scoped_ptr<RTPPayloadRegistry> payload_registry_; - scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_; - - const scoped_ptr<CriticalSectionWrapper> crit_; - unsigned int expected_bitrate_bps_ GUARDED_BY(crit_); - unsigned int start_bitrate_bps_ GUARDED_BY(crit_); - SsrcMap rtx_media_ssrcs_ GUARDED_BY(crit_); - size_t total_sent_ GUARDED_BY(crit_); - size_t padding_sent_ GUARDED_BY(crit_); - size_t rtx_media_sent_ GUARDED_BY(crit_); - int total_packets_sent_ GUARDED_BY(crit_); - int padding_packets_sent_ GUARDED_BY(crit_); - int rtx_media_packets_sent_ GUARDED_BY(crit_); -}; - -class LowRateStreamObserver : public test::DirectTransport, - public RemoteBitrateObserver, - public PacketReceiver { - public: - LowRateStreamObserver(newapi::Transport* feedback_transport, - Clock* clock, - size_t number_of_streams, - bool rtx_used) - : clock_(clock), - number_of_streams_(number_of_streams), - rtx_used_(rtx_used), - test_done_(EventWrapper::Create()), - rtp_parser_(RtpHeaderParser::Create()), - feedback_transport_(feedback_transport), - receive_stats_(ReceiveStatistics::Create(clock)), - crit_(CriticalSectionWrapper::CreateCriticalSection()), - send_stream_(NULL), - test_state_(kFirstRampup), - state_start_ms_(clock_->TimeInMilliseconds()), - interval_start_ms_(state_start_ms_), - last_remb_bps_(0), - sent_bytes_(0), - total_overuse_bytes_(0), - suspended_in_stats_(false) { - RtpRtcp::Configuration config; - config.receive_statistics = receive_stats_.get(); - feedback_transport_.Enable(); - config.outgoing_transport = &feedback_transport_; - rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config)); - rtp_rtcp_->SetREMBStatus(true); - rtp_rtcp_->SetRTCPStatus(kRtcpNonCompound); - rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset, - kTransmissionTimeOffsetExtensionId); - AbsoluteSendTimeRemoteBitrateEstimatorFactory rbe_factory; - const uint32_t kRemoteBitrateEstimatorMinBitrateBps = 10000; - remote_bitrate_estimator_.reset( - rbe_factory.Create(this, clock, kMimdControl, - kRemoteBitrateEstimatorMinBitrateBps)); - forward_transport_config_.link_capacity_kbps = - kHighBandwidthLimitBps / 1000; - forward_transport_config_.queue_length = 100; // Something large. - test::DirectTransport::SetConfig(forward_transport_config_); - test::DirectTransport::SetReceiver(this); - } +void StreamObserver::TriggerTestDone() EXCLUSIVE_LOCKS_REQUIRED(crit_) { + ReportResult("ramp-up-total-sent", total_sent_, "bytes"); + ReportResult("ramp-up-padding-sent", padding_sent_, "bytes"); + ReportResult("ramp-up-rtx-media-sent", rtx_media_sent_, "bytes"); + ReportResult("ramp-up-total-packets-sent", total_packets_sent_, "packets"); + ReportResult("ramp-up-padding-packets-sent", + padding_packets_sent_, + "packets"); + ReportResult("ramp-up-rtx-packets-sent", + rtx_media_packets_sent_, + "packets"); + ReportResult("ramp-up-time", + ramp_up_finished_ms_ - test_start_ms_, + "milliseconds"); + test_done_->Set(); +} - virtual void SetSendStream(const VideoSendStream* send_stream) { - CriticalSectionScoped lock(crit_.get()); - send_stream_ = send_stream; - } +LowRateStreamObserver::LowRateStreamObserver( + newapi::Transport* feedback_transport, + Clock* clock, + size_t number_of_streams, + bool rtx_used) + : clock_(clock), + number_of_streams_(number_of_streams), + rtx_used_(rtx_used), + test_done_(EventWrapper::Create()), + rtp_parser_(RtpHeaderParser::Create()), + feedback_transport_(feedback_transport), + receive_stats_(ReceiveStatistics::Create(clock)), + crit_(CriticalSectionWrapper::CreateCriticalSection()), + send_stream_(NULL), + test_state_(kFirstRampup), + state_start_ms_(clock_->TimeInMilliseconds()), + interval_start_ms_(state_start_ms_), + last_remb_bps_(0), + sent_bytes_(0), + total_overuse_bytes_(0), + suspended_in_stats_(false) { + RtpRtcp::Configuration config; + config.receive_statistics = receive_stats_.get(); + feedback_transport_.Enable(); + config.outgoing_transport = &feedback_transport_; + rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config)); + rtp_rtcp_->SetREMBStatus(true); + rtp_rtcp_->SetRTCPStatus(kRtcpNonCompound); + rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset, + kTransmissionTimeOffsetExtensionId); + AbsoluteSendTimeRemoteBitrateEstimatorFactory rbe_factory; + const uint32_t kRemoteBitrateEstimatorMinBitrateBps = 10000; + remote_bitrate_estimator_.reset( + rbe_factory.Create(this, clock, kMimdControl, + kRemoteBitrateEstimatorMinBitrateBps)); + forward_transport_config_.link_capacity_kbps = + kHighBandwidthLimitBps / 1000; + forward_transport_config_.queue_length = 100; // Something large. + test::DirectTransport::SetConfig(forward_transport_config_); + test::DirectTransport::SetReceiver(this); +} - virtual void OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs, - unsigned int bitrate) { - CriticalSectionScoped lock(crit_.get()); - rtp_rtcp_->SetREMBData( - bitrate, static_cast<uint8_t>(ssrcs.size()), &ssrcs[0]); - rtp_rtcp_->Process(); - last_remb_bps_ = bitrate; - } +void LowRateStreamObserver::SetSendStream(const VideoSendStream* send_stream) { + CriticalSectionScoped lock(crit_.get()); + send_stream_ = send_stream; +} - virtual bool SendRtp(const uint8_t* data, size_t length) OVERRIDE { - CriticalSectionScoped lock(crit_.get()); - sent_bytes_ += length; - int64_t now_ms = clock_->TimeInMilliseconds(); - if (now_ms > interval_start_ms_ + 1000) { // Let at least 1 second pass. - // Verify that the send rate was about right. - unsigned int average_rate_bps = static_cast<unsigned int>(sent_bytes_) * - 8 * 1000 / (now_ms - interval_start_ms_); - // TODO(holmer): Why is this failing? - // EXPECT_LT(average_rate_bps, last_remb_bps_ * 1.1); - if (average_rate_bps > last_remb_bps_ * 1.1) { - total_overuse_bytes_ += - sent_bytes_ - - last_remb_bps_ / 8 * (now_ms - interval_start_ms_) / 1000; - } - EvolveTestState(average_rate_bps); - interval_start_ms_ = now_ms; - sent_bytes_ = 0; - } - return test::DirectTransport::SendRtp(data, length); - } +void LowRateStreamObserver::OnReceiveBitrateChanged( + const std::vector<unsigned int>& ssrcs, + unsigned int bitrate) { + CriticalSectionScoped lock(crit_.get()); + rtp_rtcp_->SetREMBData( + bitrate, static_cast<uint8_t>(ssrcs.size()), &ssrcs[0]); + rtp_rtcp_->Process(); + last_remb_bps_ = bitrate; +} - virtual DeliveryStatus DeliverPacket(const uint8_t* packet, - size_t length) OVERRIDE { - CriticalSectionScoped lock(crit_.get()); - RTPHeader header; - EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length), &header)); - receive_stats_->IncomingPacket(header, length, false); - remote_bitrate_estimator_->IncomingPacket( - clock_->TimeInMilliseconds(), static_cast<int>(length - 12), header); - if (remote_bitrate_estimator_->TimeUntilNextProcess() <= 0) { - remote_bitrate_estimator_->Process(); +bool LowRateStreamObserver::SendRtp(const uint8_t* data, size_t length) { + CriticalSectionScoped lock(crit_.get()); + sent_bytes_ += length; + int64_t now_ms = clock_->TimeInMilliseconds(); + if (now_ms > interval_start_ms_ + 1000) { // Let at least 1 second pass. + // Verify that the send rate was about right. + unsigned int average_rate_bps = static_cast<unsigned int>(sent_bytes_) * + 8 * 1000 / (now_ms - interval_start_ms_); + // TODO(holmer): Why is this failing? + // EXPECT_LT(average_rate_bps, last_remb_bps_ * 1.1); + if (average_rate_bps > last_remb_bps_ * 1.1) { + total_overuse_bytes_ += + sent_bytes_ - + last_remb_bps_ / 8 * (now_ms - interval_start_ms_) / 1000; } - suspended_in_stats_ = send_stream_->GetStats().suspended; - return DELIVERY_OK; + EvolveTestState(average_rate_bps); + interval_start_ms_ = now_ms; + sent_bytes_ = 0; } + return test::DirectTransport::SendRtp(data, length); +} - virtual bool SendRtcp(const uint8_t* packet, size_t length) OVERRIDE { - return true; +PacketReceiver::DeliveryStatus LowRateStreamObserver::DeliverPacket( + const uint8_t* packet, size_t length) { + CriticalSectionScoped lock(crit_.get()); + RTPHeader header; + EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length), &header)); + receive_stats_->IncomingPacket(header, length, false); + remote_bitrate_estimator_->IncomingPacket( + clock_->TimeInMilliseconds(), static_cast<int>(length - 12), header); + if (remote_bitrate_estimator_->TimeUntilNextProcess() <= 0) { + remote_bitrate_estimator_->Process(); } + suspended_in_stats_ = send_stream_->GetStats().suspended; + return DELIVERY_OK; +} - // Produces a string similar to "1stream_nortx", depending on the values of - // number_of_streams_ and rtx_used_; - std::string GetModifierString() { - std::string str("_"); - char temp_str[5]; - sprintf(temp_str, "%i", static_cast<int>(number_of_streams_)); - str += std::string(temp_str); - str += "stream"; - str += (number_of_streams_ > 1 ? "s" : ""); - str += "_"; - str += (rtx_used_ ? "" : "no"); - str += "rtx"; - return str; - } +bool LowRateStreamObserver::SendRtcp(const uint8_t* packet, size_t length) { + return true; +} - // This method defines the state machine for the ramp up-down-up test. - void EvolveTestState(unsigned int bitrate_bps) { - int64_t now = clock_->TimeInMilliseconds(); - CriticalSectionScoped lock(crit_.get()); - assert(send_stream_ != NULL); - switch (test_state_) { - case kFirstRampup: { - EXPECT_FALSE(suspended_in_stats_); - if (bitrate_bps > kExpectedHighBitrateBps) { - // The first ramp-up has reached the target bitrate. Change the - // channel limit, and move to the next test state. - forward_transport_config_.link_capacity_kbps = - kLowBandwidthLimitBps / 1000; - test::DirectTransport::SetConfig(forward_transport_config_); - test_state_ = kLowRate; - webrtc::test::PrintResult("ramp_up_down_up", - GetModifierString(), - "first_rampup", - now - state_start_ms_, - "ms", - false); - state_start_ms_ = now; - interval_start_ms_ = now; - sent_bytes_ = 0; - } - break; +std::string LowRateStreamObserver::GetModifierString() { + std::string str("_"); + char temp_str[5]; + sprintf(temp_str, "%i", + static_cast<int>(number_of_streams_)); + str += std::string(temp_str); + str += "stream"; + str += (number_of_streams_ > 1 ? "s" : ""); + str += "_"; + str += (rtx_used_ ? "" : "no"); + str += "rtx"; + return str; +} + +void LowRateStreamObserver::EvolveTestState(unsigned int bitrate_bps) { + int64_t now = clock_->TimeInMilliseconds(); + CriticalSectionScoped lock(crit_.get()); + assert(send_stream_ != NULL); + switch (test_state_) { + case kFirstRampup: { + EXPECT_FALSE(suspended_in_stats_); + if (bitrate_bps > kExpectedHighBitrateBps) { + // The first ramp-up has reached the target bitrate. Change the + // channel limit, and move to the next test state. + forward_transport_config_.link_capacity_kbps = + kLowBandwidthLimitBps / 1000; + test::DirectTransport::SetConfig(forward_transport_config_); + test_state_ = kLowRate; + webrtc::test::PrintResult("ramp_up_down_up", + GetModifierString(), + "first_rampup", + now - state_start_ms_, + "ms", + false); + state_start_ms_ = now; + interval_start_ms_ = now; + sent_bytes_ = 0; } - case kLowRate: { - if (bitrate_bps < kExpectedLowBitrateBps && suspended_in_stats_) { - // The ramp-down was successful. Change the channel limit back to a - // high value, and move to the next test state. - forward_transport_config_.link_capacity_kbps = - kHighBandwidthLimitBps / 1000; - test::DirectTransport::SetConfig(forward_transport_config_); - test_state_ = kSecondRampup; - webrtc::test::PrintResult("ramp_up_down_up", - GetModifierString(), - "rampdown", - now - state_start_ms_, - "ms", - false); - state_start_ms_ = now; - interval_start_ms_ = now; - sent_bytes_ = 0; - } - break; + break; + } + case kLowRate: { + if (bitrate_bps < kExpectedLowBitrateBps && suspended_in_stats_) { + // The ramp-down was successful. Change the channel limit back to a + // high value, and move to the next test state. + forward_transport_config_.link_capacity_kbps = + kHighBandwidthLimitBps / 1000; + test::DirectTransport::SetConfig(forward_transport_config_); + test_state_ = kSecondRampup; + webrtc::test::PrintResult("ramp_up_down_up", + GetModifierString(), + "rampdown", + now - state_start_ms_, + "ms", + false); + state_start_ms_ = now; + interval_start_ms_ = now; + sent_bytes_ = 0; } - case kSecondRampup: { - if (bitrate_bps > kExpectedHighBitrateBps && !suspended_in_stats_) { - webrtc::test::PrintResult("ramp_up_down_up", - GetModifierString(), - "second_rampup", - now - state_start_ms_, - "ms", - false); - webrtc::test::PrintResult("ramp_up_down_up", - GetModifierString(), - "total_overuse", - total_overuse_bytes_, - "bytes", - false); - test_done_->Set(); - } - break; + break; + } + case kSecondRampup: { + if (bitrate_bps > kExpectedHighBitrateBps && !suspended_in_stats_) { + webrtc::test::PrintResult("ramp_up_down_up", + GetModifierString(), + "second_rampup", + now - state_start_ms_, + "ms", + false); + webrtc::test::PrintResult("ramp_up_down_up", + GetModifierString(), + "total_overuse", + total_overuse_bytes_, + "bytes", + false); + test_done_->Set(); } + break; } } +} - EventTypeWrapper Wait() { - return test_done_->Wait(test::CallTest::kLongTimeoutMs); - } - - private: - static const unsigned int kHighBandwidthLimitBps = 80000; - static const unsigned int kExpectedHighBitrateBps = 60000; - static const unsigned int kLowBandwidthLimitBps = 20000; - static const unsigned int kExpectedLowBitrateBps = 20000; - enum TestStates { kFirstRampup, kLowRate, kSecondRampup }; - - Clock* const clock_; - const size_t number_of_streams_; - const bool rtx_used_; - const scoped_ptr<EventWrapper> test_done_; - const scoped_ptr<RtpHeaderParser> rtp_parser_; - scoped_ptr<RtpRtcp> rtp_rtcp_; - internal::TransportAdapter feedback_transport_; - const scoped_ptr<ReceiveStatistics> receive_stats_; - scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_; - - scoped_ptr<CriticalSectionWrapper> crit_; - const VideoSendStream* send_stream_ GUARDED_BY(crit_); - FakeNetworkPipe::Config forward_transport_config_ GUARDED_BY(crit_); - TestStates test_state_ GUARDED_BY(crit_); - int64_t state_start_ms_ GUARDED_BY(crit_); - int64_t interval_start_ms_ GUARDED_BY(crit_); - unsigned int last_remb_bps_ GUARDED_BY(crit_); - size_t sent_bytes_ GUARDED_BY(crit_); - size_t total_overuse_bytes_ GUARDED_BY(crit_); - bool suspended_in_stats_ GUARDED_BY(crit_); -}; -} // namespace +EventTypeWrapper LowRateStreamObserver::Wait() { + return test_done_->Wait(test::CallTest::kLongTimeoutMs); +} -class RampUpTest : public test::CallTest { - protected: - void RunRampUpTest(bool rtx, - size_t num_streams, - unsigned int start_bitrate_bps) { - std::vector<uint32_t> ssrcs(GenerateSsrcs(num_streams, 100)); - std::vector<uint32_t> rtx_ssrcs(GenerateSsrcs(num_streams, 200)); - StreamObserver::SsrcMap rtx_ssrc_map; - if (rtx) { - for (size_t i = 0; i < ssrcs.size(); ++i) - rtx_ssrc_map[rtx_ssrcs[i]] = ssrcs[i]; - } - test::DirectTransport receiver_transport; - StreamObserver stream_observer(rtx_ssrc_map, - &receiver_transport, - Clock::GetRealTimeClock()); - - Call::Config call_config(&stream_observer); - if (start_bitrate_bps != 0) { - call_config.start_bitrate_bps = start_bitrate_bps; - stream_observer.set_start_bitrate_bps(start_bitrate_bps); - } +void RampUpTest::RunRampUpTest(bool rtx, + size_t num_streams, + unsigned int start_bitrate_bps, + const std::string& extension_type) { + std::vector<uint32_t> ssrcs(GenerateSsrcs(num_streams, 100)); + std::vector<uint32_t> rtx_ssrcs(GenerateSsrcs(num_streams, 200)); + StreamObserver::SsrcMap rtx_ssrc_map; + if (rtx) { + for (size_t i = 0; i < ssrcs.size(); ++i) + rtx_ssrc_map[rtx_ssrcs[i]] = ssrcs[i]; + } - CreateSenderCall(call_config); - CreateSendConfig(num_streams); + CreateSendConfig(num_streams); - receiver_transport.SetReceiver(sender_call_->Receiver()); + scoped_ptr<RemoteBitrateEstimatorFactory> rbe_factory; + RateControlType control_type; + if (extension_type == RtpExtension::kAbsSendTime) { + control_type = kAimdControl; + rbe_factory.reset(new AbsoluteSendTimeRemoteBitrateEstimatorFactory); + send_config_.rtp.extensions.push_back(RtpExtension( + extension_type.c_str(), kAbsSendTimeExtensionId)); + } else { + control_type = kMimdControl; + rbe_factory.reset(new RemoteBitrateEstimatorFactory); + send_config_.rtp.extensions.push_back(RtpExtension( + extension_type.c_str(), kTransmissionTimeOffsetExtensionId)); + } - if (num_streams == 1) { - video_streams_[0].target_bitrate_bps = 2000000; - video_streams_[0].max_bitrate_bps = 2000000; - } + test::DirectTransport receiver_transport; + StreamObserver stream_observer(rtx_ssrc_map, + &receiver_transport, + Clock::GetRealTimeClock(), + rbe_factory.get(), + control_type); + + Call::Config call_config(&stream_observer); + if (start_bitrate_bps != 0) { + call_config.start_bitrate_bps = start_bitrate_bps; + stream_observer.set_start_bitrate_bps(start_bitrate_bps); + } - send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs; - send_config_.rtp.ssrcs = ssrcs; - if (rtx) { - send_config_.rtp.rtx.payload_type = kSendRtxPayloadType; - send_config_.rtp.rtx.ssrcs = rtx_ssrcs; - send_config_.rtp.rtx.pad_with_redundant_payloads = true; - } - send_config_.rtp.extensions.push_back(RtpExtension( - RtpExtension::kTOffset, kTransmissionTimeOffsetExtensionId)); - - if (num_streams == 1) { - // For single stream rampup until 1mbps - stream_observer.set_expected_bitrate_bps(kSingleStreamTargetBps); - } else { - // For multi stream rampup until all streams are being sent. That means - // enough birate to send all the target streams plus the min bitrate of - // the last one. - int expected_bitrate_bps = video_streams_.back().min_bitrate_bps; - for (size_t i = 0; i < video_streams_.size() - 1; ++i) { - expected_bitrate_bps += video_streams_[i].target_bitrate_bps; - } - stream_observer.set_expected_bitrate_bps(expected_bitrate_bps); - } + CreateSenderCall(call_config); - CreateStreams(); - CreateFrameGeneratorCapturer(); + receiver_transport.SetReceiver(sender_call_->Receiver()); - Start(); + if (num_streams == 1) { + video_streams_[0].target_bitrate_bps = 2000000; + video_streams_[0].max_bitrate_bps = 2000000; + } - EXPECT_EQ(kEventSignaled, stream_observer.Wait()); + send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs; + send_config_.rtp.ssrcs = ssrcs; + if (rtx) { + send_config_.rtp.rtx.payload_type = kSendRtxPayloadType; + send_config_.rtp.rtx.ssrcs = rtx_ssrcs; + send_config_.rtp.rtx.pad_with_redundant_payloads = true; + } - Stop(); - DestroyStreams(); + if (num_streams == 1) { + // For single stream rampup until 1mbps + stream_observer.set_expected_bitrate_bps(kSingleStreamTargetBps); + } else { + // For multi stream rampup until all streams are being sent. That means + // enough birate to send all the target streams plus the min bitrate of + // the last one. + int expected_bitrate_bps = video_streams_.back().min_bitrate_bps; + for (size_t i = 0; i < video_streams_.size() - 1; ++i) { + expected_bitrate_bps += video_streams_[i].target_bitrate_bps; + } + stream_observer.set_expected_bitrate_bps(expected_bitrate_bps); } - void RunRampUpDownUpTest(size_t number_of_streams, bool rtx) { - std::vector<uint32_t> ssrcs; - for (size_t i = 0; i < number_of_streams; ++i) - ssrcs.push_back(static_cast<uint32_t>(i + 1)); - test::DirectTransport receiver_transport; - LowRateStreamObserver stream_observer( - &receiver_transport, Clock::GetRealTimeClock(), number_of_streams, rtx); + CreateStreams(); + CreateFrameGeneratorCapturer(); - Call::Config call_config(&stream_observer); - webrtc::Config webrtc_config; - call_config.webrtc_config = &webrtc_config; - webrtc_config.Set<PaddingStrategy>(new PaddingStrategy(rtx)); - CreateSenderCall(call_config); - receiver_transport.SetReceiver(sender_call_->Receiver()); + Start(); - CreateSendConfig(number_of_streams); + EXPECT_EQ(kEventSignaled, stream_observer.Wait()); - send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs; - send_config_.rtp.extensions.push_back(RtpExtension( - RtpExtension::kTOffset, kTransmissionTimeOffsetExtensionId)); - send_config_.suspend_below_min_bitrate = true; + Stop(); + DestroyStreams(); +} - CreateStreams(); - stream_observer.SetSendStream(send_stream_); +void RampUpTest::RunRampUpDownUpTest(size_t number_of_streams, bool rtx) { + test::DirectTransport receiver_transport; + LowRateStreamObserver stream_observer( + &receiver_transport, Clock::GetRealTimeClock(), number_of_streams, rtx); + + Call::Config call_config(&stream_observer); + CreateSenderCall(call_config); + receiver_transport.SetReceiver(sender_call_->Receiver()); + + CreateSendConfig(number_of_streams); + + send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs; + send_config_.rtp.extensions.push_back(RtpExtension( + RtpExtension::kTOffset, kTransmissionTimeOffsetExtensionId)); + send_config_.suspend_below_min_bitrate = true; + if (rtx) { + send_config_.rtp.rtx.payload_type = kSendRtxPayloadType; + send_config_.rtp.rtx.ssrcs = GenerateSsrcs(number_of_streams, 200); + send_config_.rtp.rtx.pad_with_redundant_payloads = true; + } - CreateFrameGeneratorCapturer(); + CreateStreams(); + stream_observer.SetSendStream(send_stream_); - Start(); + CreateFrameGeneratorCapturer(); - EXPECT_EQ(kEventSignaled, stream_observer.Wait()); + Start(); - Stop(); - DestroyStreams(); - } + EXPECT_EQ(kEventSignaled, stream_observer.Wait()); - private: - std::vector<uint32_t> GenerateSsrcs(size_t num_streams, - uint32_t ssrc_offset) { - std::vector<uint32_t> ssrcs; - for (size_t i = 0; i != num_streams; ++i) - ssrcs.push_back(static_cast<uint32_t>(ssrc_offset + i)); - return ssrcs; - } -}; + Stop(); + DestroyStreams(); +} TEST_F(RampUpTest, SingleStream) { - RunRampUpTest(false, 1, 0); + RunRampUpTest(false, 1, 0, RtpExtension::kTOffset); } TEST_F(RampUpTest, Simulcast) { - RunRampUpTest(false, 3, 0); + RunRampUpTest(false, 3, 0, RtpExtension::kTOffset); } TEST_F(RampUpTest, SimulcastWithRtx) { - RunRampUpTest(true, 3, 0); + RunRampUpTest(true, 3, 0, RtpExtension::kTOffset); } TEST_F(RampUpTest, SingleStreamWithHighStartBitrate) { - RunRampUpTest(false, 1, 0.9 * kSingleStreamTargetBps); + RunRampUpTest(false, 1, 0.9 * kSingleStreamTargetBps, RtpExtension::kTOffset); } TEST_F(RampUpTest, UpDownUpOneStream) { RunRampUpDownUpTest(1, false); } diff --git a/video/rampup_tests.h b/video/rampup_tests.h new file mode 100644 index 00000000..69399b41 --- /dev/null +++ b/video/rampup_tests.h @@ -0,0 +1,159 @@ +/* + * Copyright (c) 2014 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_VIDEO_RAMPUP_TESTS_H_ +#define WEBRTC_VIDEO_RAMPUP_TESTS_H_ + +#include <map> +#include <string> +#include <vector> + +#include "webrtc/call.h" +#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" +#include "webrtc/system_wrappers/interface/event_wrapper.h" +#include "webrtc/system_wrappers/interface/scoped_ptr.h" +#include "webrtc/test/call_test.h" +#include "webrtc/video/transport_adapter.h" + +namespace webrtc { + +static const int kTransmissionTimeOffsetExtensionId = 6; +static const int kAbsSendTimeExtensionId = 7; +static const unsigned int kSingleStreamTargetBps = 1000000; + +class Clock; +class CriticalSectionWrapper; +class ReceiveStatistics; +class RtpHeaderParser; +class RTPPayloadRegistry; +class RtpRtcp; + +class StreamObserver : public newapi::Transport, public RemoteBitrateObserver { + public: + typedef std::map<uint32_t, int> BytesSentMap; + typedef std::map<uint32_t, uint32_t> SsrcMap; + StreamObserver(const SsrcMap& rtx_media_ssrcs, + newapi::Transport* feedback_transport, + Clock* clock, + RemoteBitrateEstimatorFactory* rbe_factory, + RateControlType control_type); + + void set_expected_bitrate_bps(unsigned int expected_bitrate_bps); + + void set_start_bitrate_bps(unsigned int start_bitrate_bps); + + virtual void OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs, + unsigned int bitrate) OVERRIDE; + + virtual bool SendRtp(const uint8_t* packet, size_t length) OVERRIDE; + + virtual bool SendRtcp(const uint8_t* packet, size_t length) OVERRIDE; + + EventTypeWrapper Wait(); + + private: + void ReportResult(const std::string& measurement, + size_t value, + const std::string& units); + void TriggerTestDone() EXCLUSIVE_LOCKS_REQUIRED(crit_); + + Clock* const clock_; + const scoped_ptr<EventWrapper> test_done_; + const scoped_ptr<RtpHeaderParser> rtp_parser_; + scoped_ptr<RtpRtcp> rtp_rtcp_; + internal::TransportAdapter feedback_transport_; + const scoped_ptr<ReceiveStatistics> receive_stats_; + const scoped_ptr<RTPPayloadRegistry> payload_registry_; + scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_; + + const scoped_ptr<CriticalSectionWrapper> crit_; + unsigned int expected_bitrate_bps_ GUARDED_BY(crit_); + unsigned int start_bitrate_bps_ GUARDED_BY(crit_); + SsrcMap rtx_media_ssrcs_ GUARDED_BY(crit_); + size_t total_sent_ GUARDED_BY(crit_); + size_t padding_sent_ GUARDED_BY(crit_); + size_t rtx_media_sent_ GUARDED_BY(crit_); + int total_packets_sent_ GUARDED_BY(crit_); + int padding_packets_sent_ GUARDED_BY(crit_); + int rtx_media_packets_sent_ GUARDED_BY(crit_); + int64_t test_start_ms_ GUARDED_BY(crit_); + int64_t ramp_up_finished_ms_ GUARDED_BY(crit_); +}; + +class LowRateStreamObserver : public test::DirectTransport, + public RemoteBitrateObserver, + public PacketReceiver { + public: + LowRateStreamObserver(newapi::Transport* feedback_transport, + Clock* clock, + size_t number_of_streams, + bool rtx_used); + + virtual void SetSendStream(const VideoSendStream* send_stream); + + virtual void OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs, + unsigned int bitrate); + + virtual bool SendRtp(const uint8_t* data, size_t length) OVERRIDE; + + virtual DeliveryStatus DeliverPacket(const uint8_t* packet, + size_t length) OVERRIDE; + + virtual bool SendRtcp(const uint8_t* packet, size_t length) OVERRIDE; + + // Produces a string similar to "1stream_nortx", depending on the values of + // number_of_streams_ and rtx_used_; + std::string GetModifierString(); + + // This method defines the state machine for the ramp up-down-up test. + void EvolveTestState(unsigned int bitrate_bps); + + EventTypeWrapper Wait(); + + private: + static const unsigned int kHighBandwidthLimitBps = 80000; + static const unsigned int kExpectedHighBitrateBps = 60000; + static const unsigned int kLowBandwidthLimitBps = 20000; + static const unsigned int kExpectedLowBitrateBps = 20000; + enum TestStates { kFirstRampup, kLowRate, kSecondRampup }; + + Clock* const clock_; + const size_t number_of_streams_; + const bool rtx_used_; + const scoped_ptr<EventWrapper> test_done_; + const scoped_ptr<RtpHeaderParser> rtp_parser_; + scoped_ptr<RtpRtcp> rtp_rtcp_; + internal::TransportAdapter feedback_transport_; + const scoped_ptr<ReceiveStatistics> receive_stats_; + scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_; + + scoped_ptr<CriticalSectionWrapper> crit_; + const VideoSendStream* send_stream_ GUARDED_BY(crit_); + FakeNetworkPipe::Config forward_transport_config_ GUARDED_BY(crit_); + TestStates test_state_ GUARDED_BY(crit_); + int64_t state_start_ms_ GUARDED_BY(crit_); + int64_t interval_start_ms_ GUARDED_BY(crit_); + unsigned int last_remb_bps_ GUARDED_BY(crit_); + size_t sent_bytes_ GUARDED_BY(crit_); + size_t total_overuse_bytes_ GUARDED_BY(crit_); + bool suspended_in_stats_ GUARDED_BY(crit_); +}; + +class RampUpTest : public test::CallTest { + protected: + void RunRampUpTest(bool rtx, + size_t num_streams, + unsigned int start_bitrate_bps, + const std::string& extension_type); + + void RunRampUpDownUpTest(size_t number_of_streams, bool rtx); +}; +} // namespace webrtc +#endif // WEBRTC_VIDEO_RAMPUP_TESTS_H_ diff --git a/video/video_send_stream.cc b/video/video_send_stream.cc index 4d32ab40..ff3fe508 100644 --- a/video/video_send_stream.cc +++ b/video/video_send_stream.cc @@ -108,18 +108,21 @@ std::string VideoSendStream::Config::ToString() const { } namespace internal { -VideoSendStream::VideoSendStream(newapi::Transport* transport, - CpuOveruseObserver* overuse_observer, - webrtc::VideoEngine* video_engine, - const VideoSendStream::Config& config, - const std::vector<VideoStream> video_streams, - const void* encoder_settings, - int base_channel, - int start_bitrate_bps) +VideoSendStream::VideoSendStream( + newapi::Transport* transport, + CpuOveruseObserver* overuse_observer, + webrtc::VideoEngine* video_engine, + const VideoSendStream::Config& config, + const std::vector<VideoStream> video_streams, + const void* encoder_settings, + const std::map<uint32_t, RtpState>& suspended_ssrcs, + int base_channel, + int start_bitrate_bps) : transport_adapter_(transport), encoded_frame_proxy_(config.post_encode_callback), config_(config), start_bitrate_bps_(start_bitrate_bps), + suspended_ssrcs_(suspended_ssrcs), external_codec_(NULL), channel_(-1), stats_proxy_(new SendStatisticsProxy(config, this)) { @@ -403,6 +406,9 @@ void VideoSendStream::ConfigureSsrcs() { uint32_t ssrc = config_.rtp.ssrcs[i]; rtp_rtcp_->SetLocalSSRC( channel_, ssrc, kViEStreamTypeNormal, static_cast<unsigned char>(i)); + RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc); + if (it != suspended_ssrcs_.end()) + rtp_rtcp_->SetRtpStateForSsrc(channel_, ssrc, it->second); } if (config_.rtp.rtx.ssrcs.empty()) { @@ -412,11 +418,15 @@ void VideoSendStream::ConfigureSsrcs() { // Set up RTX. assert(config_.rtp.rtx.ssrcs.size() == config_.rtp.ssrcs.size()); - for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) { + for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) { + uint32_t ssrc = config_.rtp.rtx.ssrcs[i]; rtp_rtcp_->SetLocalSSRC(channel_, config_.rtp.rtx.ssrcs[i], kViEStreamTypeRtx, static_cast<unsigned char>(i)); + RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc); + if (it != suspended_ssrcs_.end()) + rtp_rtcp_->SetRtpStateForSsrc(channel_, ssrc, it->second); } if (config_.rtp.rtx.pad_with_redundant_payloads) { @@ -427,5 +437,20 @@ void VideoSendStream::ConfigureSsrcs() { rtp_rtcp_->SetRtxSendPayloadType(channel_, config_.rtp.rtx.payload_type); } +std::map<uint32_t, RtpState> VideoSendStream::GetRtpStates() const { + std::map<uint32_t, RtpState> rtp_states; + for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) { + uint32_t ssrc = config_.rtp.ssrcs[i]; + rtp_states[ssrc] = rtp_rtcp_->GetRtpStateForSsrc(channel_, ssrc); + } + + for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) { + uint32_t ssrc = config_.rtp.rtx.ssrcs[i]; + rtp_states[ssrc] = rtp_rtcp_->GetRtpStateForSsrc(channel_, ssrc); + } + + return rtp_states; +} + } // namespace internal } // namespace webrtc diff --git a/video/video_send_stream.h b/video/video_send_stream.h index df65b74e..e1770624 100644 --- a/video/video_send_stream.h +++ b/video/video_send_stream.h @@ -11,7 +11,10 @@ #ifndef WEBRTC_VIDEO_VIDEO_SEND_STREAM_H_ #define WEBRTC_VIDEO_VIDEO_SEND_STREAM_H_ +#include <map> + #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" +#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" #include "webrtc/video/encoded_frame_callback_adapter.h" #include "webrtc/video/send_statistics_proxy.h" #include "webrtc/video/transport_adapter.h" @@ -44,6 +47,7 @@ class VideoSendStream : public webrtc::VideoSendStream, const VideoSendStream::Config& config, const std::vector<VideoStream> video_streams, const void* encoder_settings, + const std::map<uint32_t, RtpState>& suspended_ssrcs, int base_channel, int start_bitrate); @@ -65,6 +69,9 @@ class VideoSendStream : public webrtc::VideoSendStream, // From webrtc::VideoSendStream. virtual VideoSendStreamInput* Input() OVERRIDE; + typedef std::map<uint32_t, RtpState> RtpStateMap; + RtpStateMap GetRtpStates() const; + protected: // From SendStatisticsProxy::StreamStatsProvider. virtual bool GetSendSideDelay(VideoSendStream::Stats* stats) OVERRIDE; @@ -76,6 +83,7 @@ class VideoSendStream : public webrtc::VideoSendStream, EncodedFrameCallbackAdapter encoded_frame_proxy_; const VideoSendStream::Config config_; const int start_bitrate_bps_; + std::map<uint32_t, RtpState> suspended_ssrcs_; ViEBase* video_engine_base_; ViECapture* capture_; diff --git a/video/video_send_stream_tests.cc b/video/video_send_stream_tests.cc index b1197ef6..08c4fafa 100644 --- a/video/video_send_stream_tests.cc +++ b/video/video_send_stream_tests.cc @@ -137,7 +137,7 @@ TEST_F(VideoSendStreamTest, SupportsAbsoluteSendTime) { virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { RTPHeader header; - EXPECT_TRUE(parser_->Parse(packet, static_cast<int>(length), &header)); + EXPECT_TRUE(parser_->Parse(packet, length, &header)); EXPECT_FALSE(header.extension.hasTransmissionTimeOffset); EXPECT_TRUE(header.extension.hasAbsoluteSendTime); @@ -178,7 +178,7 @@ TEST_F(VideoSendStreamTest, SupportsTransmissionTimeOffset) { private: virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { RTPHeader header; - EXPECT_TRUE(parser_->Parse(packet, static_cast<int>(length), &header)); + EXPECT_TRUE(parser_->Parse(packet, length, &header)); EXPECT_TRUE(header.extension.hasTransmissionTimeOffset); EXPECT_FALSE(header.extension.hasAbsoluteSendTime); @@ -323,7 +323,7 @@ TEST_F(VideoSendStreamTest, SupportsFec) { private: virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { RTPHeader header; - EXPECT_TRUE(parser_->Parse(packet, static_cast<int>(length), &header)); + EXPECT_TRUE(parser_->Parse(packet, length, &header)); // Send lossy receive reports to trigger FEC enabling. if (send_count_++ % 2 != 0) { @@ -398,7 +398,7 @@ void VideoSendStreamTest::TestNackRetransmission( private: virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { RTPHeader header; - EXPECT_TRUE(parser_->Parse(packet, static_cast<int>(length), &header)); + EXPECT_TRUE(parser_->Parse(packet, length, &header)); // Nack second packet after receiving the third one. if (++send_count_ == 3) { @@ -470,7 +470,7 @@ TEST_F(VideoSendStreamTest, RetransmitsNack) { TEST_F(VideoSendStreamTest, RetransmitsNackOverRtx) { // NACKs over RTX should use a separate SSRC. - TestNackRetransmission(kSendRtxSsrc, kSendRtxPayloadType); + TestNackRetransmission(kSendRtxSsrcs[0], kSendRtxPayloadType); } void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format, @@ -724,7 +724,7 @@ TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) { CriticalSectionScoped lock(crit_.get()); ++rtp_count_; RTPHeader header; - EXPECT_TRUE(parser_->Parse(packet, static_cast<int>(length), &header)); + EXPECT_TRUE(parser_->Parse(packet, length, &header)); last_sequence_number_ = header.sequenceNumber; if (test_state_ == kBeforeSuspend) { @@ -1041,11 +1041,11 @@ TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) { private: virtual DeliveryStatus DeliverPacket(const uint8_t* packet, size_t length) OVERRIDE { - if (RtpHeaderParser::IsRtcp(packet, static_cast<int>(length))) + if (RtpHeaderParser::IsRtcp(packet, length)) return DELIVERY_OK; RTPHeader header; - if (!parser_->Parse(packet, static_cast<int>(length), &header)) + if (!parser_->Parse(packet, length, &header)) return DELIVERY_PACKET_ERROR; assert(stream_ != NULL); VideoSendStream::Stats stats = stream_->GetStats(); @@ -1244,4 +1244,129 @@ I420VideoFrame* CreateI420VideoFrame(int width, int height, uint8_t data) { return frame; } +TEST_F(VideoSendStreamTest, EncoderIsProperlyInitializedAndDestroyed) { + class EncoderStateObserver : public test::SendTest, public VideoEncoder { + public: + EncoderStateObserver() + : SendTest(kDefaultTimeoutMs), + crit_(CriticalSectionWrapper::CreateCriticalSection()), + initialized_(false), + callback_registered_(false), + num_releases_(0), + released_(false) {} + + bool IsReleased() { + CriticalSectionScoped lock(crit_.get()); + return released_; + } + + bool IsReadyForEncode() { + CriticalSectionScoped lock(crit_.get()); + return initialized_ && callback_registered_; + } + + size_t num_releases() { + CriticalSectionScoped lock(crit_.get()); + return num_releases_; + } + + private: + virtual int32_t InitEncode(const VideoCodec* codecSettings, + int32_t numberOfCores, + uint32_t maxPayloadSize) OVERRIDE { + CriticalSectionScoped lock(crit_.get()); + EXPECT_FALSE(initialized_); + initialized_ = true; + released_ = false; + return 0; + } + + virtual int32_t Encode( + const I420VideoFrame& inputImage, + const CodecSpecificInfo* codecSpecificInfo, + const std::vector<VideoFrameType>* frame_types) OVERRIDE { + EXPECT_TRUE(IsReadyForEncode()); + + observation_complete_->Set(); + return 0; + } + + virtual int32_t RegisterEncodeCompleteCallback( + EncodedImageCallback* callback) OVERRIDE { + CriticalSectionScoped lock(crit_.get()); + EXPECT_TRUE(initialized_); + callback_registered_ = true; + return 0; + } + + virtual int32_t Release() OVERRIDE { + CriticalSectionScoped lock(crit_.get()); + EXPECT_TRUE(IsReadyForEncode()); + EXPECT_FALSE(released_); + initialized_ = false; + callback_registered_ = false; + released_ = true; + ++num_releases_; + return 0; + } + + virtual int32_t SetChannelParameters(uint32_t packetLoss, + int rtt) OVERRIDE { + EXPECT_TRUE(IsReadyForEncode()); + return 0; + } + + virtual int32_t SetRates(uint32_t newBitRate, uint32_t frameRate) OVERRIDE { + EXPECT_TRUE(IsReadyForEncode()); + return 0; + } + + virtual void OnStreamsCreated( + VideoSendStream* send_stream, + const std::vector<VideoReceiveStream*>& receive_streams) OVERRIDE { + // Encoder initialization should be done in stream construction before + // starting. + EXPECT_TRUE(IsReadyForEncode()); + stream_ = send_stream; + } + + virtual void ModifyConfigs( + VideoSendStream::Config* send_config, + std::vector<VideoReceiveStream::Config>* receive_configs, + std::vector<VideoStream>* video_streams) OVERRIDE { + send_config->encoder_settings.encoder = this; + video_streams_ = *video_streams; + } + + virtual void PerformTest() OVERRIDE { + EXPECT_EQ(kEventSignaled, Wait()) + << "Timed out while waiting for Encode."; + EXPECT_EQ(0u, num_releases()); + stream_->ReconfigureVideoEncoder(video_streams_, NULL); + EXPECT_EQ(0u, num_releases()); + stream_->Stop(); + // Encoder should not be released before destroying the VideoSendStream. + EXPECT_FALSE(IsReleased()); + EXPECT_TRUE(IsReadyForEncode()); + stream_->Start(); + // Sanity check, make sure we still encode frames with this encoder. + EXPECT_EQ(kEventSignaled, Wait()) + << "Timed out while waiting for Encode."; + } + + scoped_ptr<CriticalSectionWrapper> crit_; + VideoSendStream* stream_; + bool initialized_ GUARDED_BY(crit_); + bool callback_registered_ GUARDED_BY(crit_); + size_t num_releases_ GUARDED_BY(crit_); + bool released_ GUARDED_BY(crit_); + std::vector<VideoStream> video_streams_; + } test_encoder; + + RunBaseTest(&test_encoder); + + EXPECT_TRUE(test_encoder.IsReleased()); + EXPECT_EQ(1u, test_encoder.num_releases()); +} + } // namespace webrtc diff --git a/video_engine/include/vie_base.h b/video_engine/include/vie_base.h index 0a528cb1..f4b99ae7 100644 --- a/video_engine/include/vie_base.h +++ b/video_engine/include/vie_base.h @@ -43,29 +43,11 @@ class CpuOveruseObserver { virtual ~CpuOveruseObserver() {} }; -// Limits on standard deviation for under/overuse. -#ifdef WEBRTC_ANDROID -const float kOveruseStdDevMs = 32.0f; -const float kNormalUseStdDevMs = 27.0f; -#elif WEBRTC_LINUX -const float kOveruseStdDevMs = 20.0f; -const float kNormalUseStdDevMs = 14.0f; -#elif WEBRTC_MAC -const float kOveruseStdDevMs = 27.0f; -const float kNormalUseStdDevMs = 21.0f; -#elif WEBRTC_WIN -const float kOveruseStdDevMs = 20.0f; -const float kNormalUseStdDevMs = 14.0f; -#else -const float kOveruseStdDevMs = 30.0f; -const float kNormalUseStdDevMs = 20.0f; -#endif - struct CpuOveruseOptions { CpuOveruseOptions() : enable_capture_jitter_method(true), - low_capture_jitter_threshold_ms(kNormalUseStdDevMs), - high_capture_jitter_threshold_ms(kOveruseStdDevMs), + low_capture_jitter_threshold_ms(20.0f), + high_capture_jitter_threshold_ms(30.0f), enable_encode_usage_method(false), low_encode_usage_threshold_percent(60), high_encode_usage_threshold_percent(90), @@ -216,26 +198,11 @@ class WEBRTC_DLLEXPORT ViEBase { CpuOveruseObserver* observer) = 0; // Sets options for cpu overuse detector. - // TODO(asapersson): Remove default implementation. virtual int SetCpuOveruseOptions(int channel, - const CpuOveruseOptions& options) { - return -1; - } + const CpuOveruseOptions& options) = 0; // Gets cpu overuse measures. - // TODO(asapersson): Remove default implementation. - virtual int GetCpuOveruseMetrics(int channel, - CpuOveruseMetrics* metrics) { - return -1; - } - // TODO(asapersson): Remove this function when libjingle has been updated. - virtual int CpuOveruseMeasures(int channel, - int* capture_jitter_ms, - int* avg_encode_time_ms, - int* encode_usage_percent, - int* capture_queue_delay_ms_per_s) { - return -1; - } + virtual int GetCpuOveruseMetrics(int channel, CpuOveruseMetrics* metrics) = 0; // Specifies the VoiceEngine and VideoEngine channel pair to use for // audio/video synchronization. diff --git a/video_engine/include/vie_rtp_rtcp.h b/video_engine/include/vie_rtp_rtcp.h index 972ca584..3a565610 100644 --- a/video_engine/include/vie_rtp_rtcp.h +++ b/video_engine/include/vie_rtp_rtcp.h @@ -23,6 +23,7 @@ #define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_RTP_RTCP_H_ #include "webrtc/common_types.h" +#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" namespace webrtc { @@ -151,6 +152,17 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP { virtual int SetStartSequenceNumber(const int video_channel, unsigned short sequence_number) = 0; + // TODO(pbos): Remove default implementation once this has been implemented + // in libjingle. + virtual void SetRtpStateForSsrc(int video_channel, + uint32_t ssrc, + const RtpState& rtp_state) {} + // TODO(pbos): Remove default implementation once this has been implemented + // in libjingle. + virtual RtpState GetRtpStateForSsrc(int video_channel, uint32_t ssrc) { + return RtpState(); + } + // This function sets the RTCP status for the specified channel. // Default mode is kRtcpCompound_RFC4585. virtual int SetRTCPStatus(const int video_channel, @@ -261,8 +273,7 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP { // Enables/disables RTCP Receiver Reference Time Report Block extension/ // DLRR Report Block extension (RFC 3611). - // TODO(asapersson): Remove default implementation. - virtual int SetRtcpXrRrtrStatus(int video_channel, bool enable) { return -1; } + virtual int SetRtcpXrRrtrStatus(int video_channel, bool enable) = 0; // Enables transmission smoothening, i.e. packets belonging to the same frame // will be sent over a longer period of time instead of sending them diff --git a/video_engine/test/auto_test/automated/vie_network_test.cc b/video_engine/test/auto_test/automated/vie_network_test.cc index 1e46b3d0..a2d060e7 100644 --- a/video_engine/test/auto_test/automated/vie_network_test.cc +++ b/video_engine/test/auto_test/automated/vie_network_test.cc @@ -31,7 +31,7 @@ class RtcpCollectorTransport : public webrtc::Transport { } virtual int SendRTCPPacket(int channel, const void* data, int len) { const uint8_t* buf = static_cast<const uint8_t*>(data); - webrtc::ModuleRTPUtility::RTPHeaderParser parser(buf, len); + webrtc::RtpUtility::RtpHeaderParser parser(buf, len); if (parser.RTCP()) { Packet p; p.channel = channel; diff --git a/video_engine/test/auto_test/source/vie_autotest.cc b/video_engine/test/auto_test/source/vie_autotest.cc index 188567cb..fb1a46f2 100644 --- a/video_engine/test/auto_test/source/vie_autotest.cc +++ b/video_engine/test/auto_test/source/vie_autotest.cc @@ -101,6 +101,9 @@ void ViEAutoTest::PrintVideoCodec(const webrtc::VideoCodec videoCodec) case webrtc::kVideoCodecI420: ViETest::Log("\tcodecType: I420"); break; + case webrtc::kVideoCodecH264: + ViETest::Log("\tcodecType: H264"); + break; case webrtc::kVideoCodecRED: ViETest::Log("\tcodecType: RED"); break; diff --git a/video_engine/vie_base_impl.cc b/video_engine/vie_base_impl.cc index 8b4da279..6ba2fd5f 100644 --- a/video_engine/vie_base_impl.cc +++ b/video_engine/vie_base_impl.cc @@ -121,37 +121,6 @@ int ViEBaseImpl::SetCpuOveruseOptions(int video_channel, return -1; } -int ViEBaseImpl::CpuOveruseMeasures(int video_channel, - int* capture_jitter_ms, - int* avg_encode_time_ms, - int* encode_usage_percent, - int* capture_queue_delay_ms_per_s) { - ViEChannelManagerScoped cs(*(shared_data_.channel_manager())); - ViEChannel* vie_channel = cs.Channel(video_channel); - if (!vie_channel) { - shared_data_.SetLastError(kViEBaseInvalidChannelId); - return -1; - } - ViEEncoder* vie_encoder = cs.Encoder(video_channel); - assert(vie_encoder); - - ViEInputManagerScoped is(*(shared_data_.input_manager())); - ViEFrameProviderBase* provider = is.FrameProvider(vie_encoder); - if (provider) { - ViECapturer* capturer = is.Capture(provider->Id()); - if (capturer) { - CpuOveruseMetrics metrics; - capturer->GetCpuOveruseMetrics(&metrics); - *capture_jitter_ms = metrics.capture_jitter_ms; - *avg_encode_time_ms = metrics.avg_encode_time_ms; - *encode_usage_percent = metrics.encode_usage_percent; - *capture_queue_delay_ms_per_s = metrics.capture_queue_delay_ms_per_s; - return 0; - } - } - return -1; -} - int ViEBaseImpl::GetCpuOveruseMetrics(int video_channel, CpuOveruseMetrics* metrics) { ViEChannelManagerScoped cs(*(shared_data_.channel_manager())); diff --git a/video_engine/vie_base_impl.h b/video_engine/vie_base_impl.h index d6a046e6..2f847bc6 100644 --- a/video_engine/vie_base_impl.h +++ b/video_engine/vie_base_impl.h @@ -37,11 +37,6 @@ class ViEBaseImpl const CpuOveruseOptions& options); virtual int GetCpuOveruseMetrics(int channel, CpuOveruseMetrics* metrics); - virtual int CpuOveruseMeasures(int channel, - int* capture_jitter_ms, - int* avg_encode_time_ms, - int* encode_usage_percent, - int* capture_queue_delay_ms_per_s); virtual int CreateChannel(int& video_channel); // NOLINT virtual int CreateChannel(int& video_channel, // NOLINT const Config* config); diff --git a/video_engine/vie_channel.cc b/video_engine/vie_channel.cc index 80d3065b..46622518 100644 --- a/video_engine/vie_channel.cc +++ b/video_engine/vie_channel.cc @@ -116,6 +116,7 @@ ViEChannel::ViEChannel(int32_t channel_id, configuration.remote_bitrate_estimator = remote_bitrate_estimator; configuration.paced_sender = paced_sender; configuration.receive_statistics = vie_receiver_.GetReceiveStatistics(); + configuration.send_bitrate_observer = &send_bitrate_observer_; rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(configuration)); vie_receiver_.SetRtpRtcpModule(rtp_rtcp_.get()); @@ -278,6 +279,11 @@ int32_t ViEChannel::SetSendCodec(const VideoCodec& video_codec, rtp_rtcp->SetSendingStatus(rtp_rtcp_->Sending()); rtp_rtcp->SetSendingMediaStatus(rtp_rtcp_->SendingMedia()); + int mode; + uint32_t ssrc; + int payload_type; + rtp_rtcp_->RTXSendStatus(&mode, &ssrc, &payload_type); + rtp_rtcp->SetRTXSendStatus(mode); simulcast_rtp_rtcp_.push_back(rtp_rtcp); // Silently ignore error. @@ -295,7 +301,6 @@ int32_t ViEChannel::SetSendCodec(const VideoCodec& video_codec, rtp_rtcp->RegisterSendFrameCountObserver(NULL); rtp_rtcp->RegisterSendChannelRtcpStatisticsCallback(NULL); rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(NULL); - rtp_rtcp->RegisterVideoBitrateObserver(NULL); simulcast_rtp_rtcp_.pop_back(); removed_rtp_rtcp_.push_front(rtp_rtcp); } @@ -347,8 +352,6 @@ int32_t ViEChannel::SetSendCodec(const VideoCodec& video_codec, rtp_rtcp_->GetSendChannelRtcpStatisticsCallback()); rtp_rtcp->RegisterSendChannelRtpStatisticsCallback( rtp_rtcp_->GetSendChannelRtpStatisticsCallback()); - rtp_rtcp->RegisterVideoBitrateObserver( - rtp_rtcp_->GetVideoBitrateObserver()); } // |RegisterSimulcastRtpRtcpModules| resets all old weak pointers and old // modules can be deleted after this step. @@ -362,7 +365,6 @@ int32_t ViEChannel::SetSendCodec(const VideoCodec& video_codec, rtp_rtcp->RegisterSendFrameCountObserver(NULL); rtp_rtcp->RegisterSendChannelRtcpStatisticsCallback(NULL); rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(NULL); - rtp_rtcp->RegisterVideoBitrateObserver(NULL); simulcast_rtp_rtcp_.pop_back(); removed_rtp_rtcp_.push_front(rtp_rtcp); } @@ -888,6 +890,21 @@ int32_t ViEChannel::SetStartSequenceNumber(uint16_t sequence_number) { return rtp_rtcp_->SetSequenceNumber(sequence_number); } +void ViEChannel::SetRtpStateForSsrc(uint32_t ssrc, const RtpState& rtp_state) { + assert(!rtp_rtcp_->Sending()); + default_rtp_rtcp_->SetRtpStateForSsrc(ssrc, rtp_state); +} + +RtpState ViEChannel::GetRtpStateForSsrc(uint32_t ssrc) { + assert(!rtp_rtcp_->Sending()); + + RtpState rtp_state; + if (!default_rtp_rtcp_->GetRtpStateForSsrc(ssrc, &rtp_state)) { + LOG(LS_ERROR) << "Couldn't get RTP state for ssrc: " << ssrc; + } + return rtp_state; +} + int32_t ViEChannel::SetRTCPCName(const char rtcp_cname[]) { if (rtp_rtcp_->Sending()) { return -1; @@ -1192,13 +1209,7 @@ bool ViEChannel::GetSendSideDelay(int* avg_send_delay, void ViEChannel::RegisterSendBitrateObserver( BitrateStatisticsObserver* observer) { - rtp_rtcp_->RegisterVideoBitrateObserver(observer); - CriticalSectionScoped cs(rtp_rtcp_cs_.get()); - for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin(); - it != simulcast_rtp_rtcp_.end(); - it++) { - (*it)->RegisterVideoBitrateObserver(observer); - } + send_bitrate_observer_.Set(observer); } void ViEChannel::GetReceiveBandwidthEstimatorStats( @@ -1516,7 +1527,6 @@ void ViEChannel::ReserveRtpRtcpModules(size_t num_modules) { rtp_rtcp->RegisterSendFrameCountObserver(NULL); rtp_rtcp->RegisterSendChannelRtcpStatisticsCallback(NULL); rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(NULL); - rtp_rtcp->RegisterVideoBitrateObserver(NULL); removed_rtp_rtcp_.push_back(rtp_rtcp); } } diff --git a/video_engine/vie_channel.h b/video_engine/vie_channel.h index c76d1298..39f9b75c 100644 --- a/video_engine/vie_channel.h +++ b/video_engine/vie_channel.h @@ -16,6 +16,7 @@ #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" +#include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/system_wrappers/interface/tick_util.h" #include "webrtc/typedefs.h" @@ -152,6 +153,9 @@ class ViEChannel // Sets the starting sequence number, must be called before StartSend. int32_t SetStartSequenceNumber(uint16_t sequence_number); + void SetRtpStateForSsrc(uint32_t ssrc, const RtpState& rtp_state); + RtpState GetRtpStateForSsrc(uint32_t ssrc); + // Sets the CName for the outgoing stream on the channel. int32_t SetRTCPCName(const char rtcp_cname[]); @@ -379,6 +383,43 @@ class ViEChannel int GetRequiredNackListSize(int target_delay_ms); void SetRtxSendStatus(bool enable); + // ViEChannel exposes methods that allow to modify observers and callbacks + // to be modified. Such an API-style is cumbersome to implement and maintain + // at all the levels when comparing to only setting them at construction. As + // so this class instantiates its children with a wrapper that can be modified + // at a later time. + template <class T> + class RegisterableCallback : public T { + public: + RegisterableCallback() + : critsect_(CriticalSectionWrapper::CreateCriticalSection()), + callback_(NULL) {} + + void Set(T* callback) { + CriticalSectionScoped cs(critsect_.get()); + callback_ = callback; + } + + protected: + // Note: this should be implemented with a RW-lock to allow simultaneous + // calls into the callback. However that doesn't seem to be needed for the + // current type of callbacks covered by this class. + scoped_ptr<CriticalSectionWrapper> critsect_; + T* callback_ GUARDED_BY(critsect_); + + private: + DISALLOW_COPY_AND_ASSIGN(RegisterableCallback); + }; + + class : public RegisterableCallback<BitrateStatisticsObserver> { + virtual void Notify(const BitrateStatistics& stats, uint32_t ssrc) { + CriticalSectionScoped cs(critsect_.get()); + if (callback_) + callback_->Notify(stats, ssrc); + } + } + send_bitrate_observer_; + int32_t channel_id_; int32_t engine_id_; uint32_t number_of_cores_; diff --git a/video_engine/vie_codec_impl.cc b/video_engine/vie_codec_impl.cc index 3ba56de5..050958e2 100644 --- a/video_engine/vie_codec_impl.cc +++ b/video_engine/vie_codec_impl.cc @@ -69,6 +69,18 @@ static void LogCodec(const VideoCodec& codec) { << ", qp max " << codec.simulcastStream[idx].qpMax; } + } else if (codec.codecType == kVideoCodecH264) { + LOG(LS_INFO) << "H264 specific settings"; + LOG(LS_INFO) << "profile: " + << codec.codecSpecific.H264.profile + << ", framedropping: " + << codec.codecSpecific.H264.frameDroppingOn + << ", keyFrameInterval: " + << codec.codecSpecific.H264.keyFrameInterval + << ", spslen: " + << codec.codecSpecific.H264.spsLen + << ", ppslen: " + << codec.codecSpecific.H264.ppsLen; } } @@ -629,7 +641,9 @@ bool ViECodecImpl::CodecValid(const VideoCodec& video_codec) { } else if ((video_codec.codecType == kVideoCodecVP8 && strncmp(video_codec.plName, "VP8", 4) == 0) || (video_codec.codecType == kVideoCodecI420 && - strncmp(video_codec.plName, "I420", 4) == 0)) { + strncmp(video_codec.plName, "I420", 4) == 0) || + (video_codec.codecType == kVideoCodecH264 && + strncmp(video_codec.plName, "H264", 4) == 0)) { // OK. } else if (video_codec.codecType != kVideoCodecGeneric) { LOG(LS_ERROR) << "Codec type and name mismatch."; diff --git a/video_engine/vie_encoder.cc b/video_engine/vie_encoder.cc index a009c06a..fd3f0d78 100644 --- a/video_engine/vie_encoder.cc +++ b/video_engine/vie_encoder.cc @@ -23,6 +23,7 @@ #include "webrtc/modules/video_coding/main/interface/video_coding.h" #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" #include "webrtc/modules/video_coding/main/source/encoded_frame.h" +#include "webrtc/system_wrappers/interface/clock.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/logging.h" #include "webrtc/system_wrappers/interface/tick_util.h" @@ -34,16 +35,6 @@ namespace webrtc { -// Pace in kbits/s until we receive first estimate. -static const int kInitialPace = 2000; - -// Pacing-rate relative to our target send rate. -// Multiplicative factor that is applied to the target bitrate to calculate the -// number of bytes that can be transmitted per interval. -// Increasing this factor will result in lower delays in cases of bitrate -// overshoots from the encoder. -static const float kPaceMultiplier = 2.5f; - // Margin on when we pause the encoder when the pacing buffer overflows relative // to the configured buffer delay. static const float kEncoderPausePacerMargin = 2.0f; @@ -172,7 +163,8 @@ ViEEncoder::ViEEncoder(int32_t engine_id, bitrate_observer_.reset(new ViEBitrateObserver(this)); pacing_callback_.reset(new ViEPacedSenderCallback(this)); paced_sender_.reset( - new PacedSender(pacing_callback_.get(), kInitialPace, kPaceMultiplier)); + new PacedSender(Clock::GetRealTimeClock(), pacing_callback_.get(), + PacedSender::kDefaultInitialPaceKbps, 0)); } bool ViEEncoder::Init() { @@ -383,8 +375,9 @@ int32_t ViEEncoder::SetEncoder(const webrtc::VideoCodec& video_codec) { if (pad_up_to_bitrate_kbps < min_transmit_bitrate_kbps_) pad_up_to_bitrate_kbps = min_transmit_bitrate_kbps_; - paced_sender_->UpdateBitrate(kPaceMultiplier * video_codec.startBitrate, - pad_up_to_bitrate_kbps); + paced_sender_->UpdateBitrate( + PacedSender::kDefaultPaceMultiplier * video_codec.startBitrate, + pad_up_to_bitrate_kbps); return 0; } @@ -894,8 +887,9 @@ void ViEEncoder::OnNetworkChanged(const uint32_t bitrate_bps, if (pad_up_to_bitrate_kbps > bitrate_kbps) pad_up_to_bitrate_kbps = bitrate_kbps; - paced_sender_->UpdateBitrate(kPaceMultiplier * bitrate_kbps, - pad_up_to_bitrate_kbps); + paced_sender_->UpdateBitrate( + PacedSender::kDefaultPaceMultiplier * bitrate_kbps, + pad_up_to_bitrate_kbps); default_rtp_rtcp_->SetTargetSendBitrate(stream_bitrates); if (video_suspended_ == video_is_suspended) return; diff --git a/video_engine/vie_rtp_rtcp_impl.cc b/video_engine/vie_rtp_rtcp_impl.cc index 53610b4a..04f5e9b0 100644 --- a/video_engine/vie_rtp_rtcp_impl.cc +++ b/video_engine/vie_rtp_rtcp_impl.cc @@ -256,6 +256,30 @@ int ViERTP_RTCPImpl::SetStartSequenceNumber(const int video_channel, return 0; } +void ViERTP_RTCPImpl::SetRtpStateForSsrc(int video_channel, + uint32_t ssrc, + const RtpState& rtp_state) { + ViEChannelManagerScoped cs(*(shared_data_->channel_manager())); + ViEChannel* vie_channel = cs.Channel(video_channel); + if (!vie_channel) + return; + + if (vie_channel->Sending()) { + LOG_F(LS_ERROR) << "channel " << video_channel << " is already sending."; + return; + } + vie_channel->SetRtpStateForSsrc(ssrc, rtp_state); +} + +RtpState ViERTP_RTCPImpl::GetRtpStateForSsrc(int video_channel, uint32_t ssrc) { + ViEChannelManagerScoped cs(*(shared_data_->channel_manager())); + ViEChannel* vie_channel = cs.Channel(video_channel); + if (!vie_channel) + return RtpState(); + + return vie_channel->GetRtpStateForSsrc(ssrc); +} + int ViERTP_RTCPImpl::SetRTCPStatus(const int video_channel, const ViERTCPMode rtcp_mode) { LOG_F(LS_INFO) << "channel: " << video_channel diff --git a/video_engine/vie_rtp_rtcp_impl.h b/video_engine/vie_rtp_rtcp_impl.h index 5eec0efe..4afe1c58 100644 --- a/video_engine/vie_rtp_rtcp_impl.h +++ b/video_engine/vie_rtp_rtcp_impl.h @@ -46,6 +46,11 @@ class ViERTP_RTCPImpl const uint8_t payload_type); virtual int SetStartSequenceNumber(const int video_channel, uint16_t sequence_number); + virtual void SetRtpStateForSsrc(int video_channel, + uint32_t ssrc, + const RtpState& rtp_state) OVERRIDE; + virtual RtpState GetRtpStateForSsrc(int video_channel, + uint32_t ssrc) OVERRIDE; virtual int SetRTCPStatus(const int video_channel, const ViERTCPMode rtcp_mode); virtual int GetRTCPStatus(const int video_channel, diff --git a/video_engine/vie_sender.cc b/video_engine/vie_sender.cc index 349bc72f..28bf3903 100644 --- a/video_engine/vie_sender.cc +++ b/video_engine/vie_sender.cc @@ -11,6 +11,7 @@ #include "webrtc/video_engine/vie_sender.h" #include <assert.h> +#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h" #include "webrtc/modules/utility/interface/rtp_dump.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" diff --git a/video_receive_stream.h b/video_receive_stream.h index 2584e204..b1633f93 100644 --- a/video_receive_stream.h +++ b/video_receive_stream.h @@ -90,7 +90,7 @@ class VideoReceiveStream { : remote_ssrc(0), local_ssrc(0), rtcp_mode(newapi::kRtcpReducedSize), - remb(false) {} + remb(true) {} // Synchronization source (stream identifier) to be received. uint32_t remote_ssrc; diff --git a/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc b/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc index cf33adf1..a678b13a 100644 --- a/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc +++ b/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc @@ -31,7 +31,9 @@ class ExtensionVerifyTransport : public webrtc::Transport { virtual int SendPacket(int channel, const void* data, int len) { webrtc::RTPHeader header; - if (parser_->Parse(static_cast<const uint8_t*>(data), len, &header)) { + if (parser_->Parse(reinterpret_cast<const uint8_t*>(data), + static_cast<size_t>(len), + &header)) { bool ok = true; if (audio_level_id_ >= 0 && !header.extension.hasAudioLevel) { diff --git a/webrtc_examples.gyp b/webrtc_examples.gyp index 56a0e9ef..e8d1b982 100644 --- a/webrtc_examples.gyp +++ b/webrtc_examples.gyp @@ -15,7 +15,6 @@ 'type': 'loadable_module', 'dependencies': [ '<(DEPTH)/third_party/icu/icu.gyp:icuuc', - '<(webrtc_root)/modules/modules.gyp:*', '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:field_trial_default', '<(webrtc_root)/test/test.gyp:channel_transport', '<(webrtc_root)/video_engine/video_engine.gyp:video_engine_core', diff --git a/webrtc_tests.gypi b/webrtc_tests.gypi index 85a7e733..78b199f0 100644 --- a/webrtc_tests.gypi +++ b/webrtc_tests.gypi @@ -79,6 +79,7 @@ 'video/call_perf_tests.cc', 'video/full_stack.cc', 'video/rampup_tests.cc', + 'video/rampup_tests.h', ], 'dependencies': [ '<(DEPTH)/testing/gtest.gyp:gtest', |