| Index: talk/media/webrtc/webrtcvideoengine2.cc
|
| diff --git a/talk/media/webrtc/webrtcvideoengine2.cc b/talk/media/webrtc/webrtcvideoengine2.cc
|
| deleted file mode 100644
|
| index 6f04d9674a15d7fb83afbf80b2e57483fbc2d626..0000000000000000000000000000000000000000
|
| --- a/talk/media/webrtc/webrtcvideoengine2.cc
|
| +++ /dev/null
|
| @@ -1,2536 +0,0 @@
|
| -/*
|
| - * libjingle
|
| - * Copyright 2014 Google Inc.
|
| - *
|
| - * Redistribution and use in source and binary forms, with or without
|
| - * modification, are permitted provided that the following conditions are met:
|
| - *
|
| - * 1. Redistributions of source code must retain the above copyright notice,
|
| - * this list of conditions and the following disclaimer.
|
| - * 2. Redistributions in binary form must reproduce the above copyright notice,
|
| - * this list of conditions and the following disclaimer in the documentation
|
| - * and/or other materials provided with the distribution.
|
| - * 3. The name of the author may not be used to endorse or promote products
|
| - * derived from this software without specific prior written permission.
|
| - *
|
| - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
|
| - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
|
| - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
|
| - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
|
| - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
|
| - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
|
| - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
|
| - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
|
| - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
|
| - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
|
| - */
|
| -
|
| -#ifdef HAVE_WEBRTC_VIDEO
|
| -#include "talk/media/webrtc/webrtcvideoengine2.h"
|
| -
|
| -#include <algorithm>
|
| -#include <set>
|
| -#include <string>
|
| -
|
| -#include "talk/media/base/videocapturer.h"
|
| -#include "talk/media/base/videorenderer.h"
|
| -#include "talk/media/webrtc/constants.h"
|
| -#include "talk/media/webrtc/simulcast.h"
|
| -#include "talk/media/webrtc/webrtcmediaengine.h"
|
| -#include "talk/media/webrtc/webrtcvideoencoderfactory.h"
|
| -#include "talk/media/webrtc/webrtcvideoframe.h"
|
| -#include "talk/media/webrtc/webrtcvoiceengine.h"
|
| -#include "webrtc/base/buffer.h"
|
| -#include "webrtc/base/logging.h"
|
| -#include "webrtc/base/stringutils.h"
|
| -#include "webrtc/base/timeutils.h"
|
| -#include "webrtc/base/trace_event.h"
|
| -#include "webrtc/call.h"
|
| -#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
|
| -#include "webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h"
|
| -#include "webrtc/system_wrappers/include/field_trial.h"
|
| -#include "webrtc/video_decoder.h"
|
| -#include "webrtc/video_encoder.h"
|
| -
|
| -namespace cricket {
|
| -namespace {
|
| -
|
| -// Wrap cricket::WebRtcVideoEncoderFactory as a webrtc::VideoEncoderFactory.
|
| -class EncoderFactoryAdapter : public webrtc::VideoEncoderFactory {
|
| - public:
|
| - // EncoderFactoryAdapter doesn't take ownership of |factory|, which is owned
|
| - // by e.g. PeerConnectionFactory.
|
| - explicit EncoderFactoryAdapter(cricket::WebRtcVideoEncoderFactory* factory)
|
| - : factory_(factory) {}
|
| - virtual ~EncoderFactoryAdapter() {}
|
| -
|
| - // Implement webrtc::VideoEncoderFactory.
|
| - webrtc::VideoEncoder* Create() override {
|
| - return factory_->CreateVideoEncoder(webrtc::kVideoCodecVP8);
|
| - }
|
| -
|
| - void Destroy(webrtc::VideoEncoder* encoder) override {
|
| - return factory_->DestroyVideoEncoder(encoder);
|
| - }
|
| -
|
| - private:
|
| - cricket::WebRtcVideoEncoderFactory* const factory_;
|
| -};
|
| -
|
| -webrtc::Call::Config::BitrateConfig GetBitrateConfigForCodec(
|
| - const VideoCodec& codec) {
|
| - webrtc::Call::Config::BitrateConfig config;
|
| - int bitrate_kbps;
|
| - if (codec.GetParam(kCodecParamMinBitrate, &bitrate_kbps) &&
|
| - bitrate_kbps > 0) {
|
| - config.min_bitrate_bps = bitrate_kbps * 1000;
|
| - } else {
|
| - config.min_bitrate_bps = 0;
|
| - }
|
| - if (codec.GetParam(kCodecParamStartBitrate, &bitrate_kbps) &&
|
| - bitrate_kbps > 0) {
|
| - config.start_bitrate_bps = bitrate_kbps * 1000;
|
| - } else {
|
| - // Do not reconfigure start bitrate unless it's specified and positive.
|
| - config.start_bitrate_bps = -1;
|
| - }
|
| - if (codec.GetParam(kCodecParamMaxBitrate, &bitrate_kbps) &&
|
| - bitrate_kbps > 0) {
|
| - config.max_bitrate_bps = bitrate_kbps * 1000;
|
| - } else {
|
| - config.max_bitrate_bps = -1;
|
| - }
|
| - return config;
|
| -}
|
| -
|
| -// An encoder factory that wraps Create requests for simulcastable codec types
|
| -// with a webrtc::SimulcastEncoderAdapter. Non simulcastable codec type
|
| -// requests are just passed through to the contained encoder factory.
|
| -class WebRtcSimulcastEncoderFactory
|
| - : public cricket::WebRtcVideoEncoderFactory {
|
| - public:
|
| - // WebRtcSimulcastEncoderFactory doesn't take ownership of |factory|, which is
|
| - // owned by e.g. PeerConnectionFactory.
|
| - explicit WebRtcSimulcastEncoderFactory(
|
| - cricket::WebRtcVideoEncoderFactory* factory)
|
| - : factory_(factory) {}
|
| -
|
| - static bool UseSimulcastEncoderFactory(
|
| - const std::vector<VideoCodec>& codecs) {
|
| - // If any codec is VP8, use the simulcast factory. If asked to create a
|
| - // non-VP8 codec, we'll just return a contained factory encoder directly.
|
| - for (const auto& codec : codecs) {
|
| - if (codec.type == webrtc::kVideoCodecVP8) {
|
| - return true;
|
| - }
|
| - }
|
| - return false;
|
| - }
|
| -
|
| - webrtc::VideoEncoder* CreateVideoEncoder(
|
| - webrtc::VideoCodecType type) override {
|
| - RTC_DCHECK(factory_ != NULL);
|
| - // If it's a codec type we can simulcast, create a wrapped encoder.
|
| - if (type == webrtc::kVideoCodecVP8) {
|
| - return new webrtc::SimulcastEncoderAdapter(
|
| - new EncoderFactoryAdapter(factory_));
|
| - }
|
| - webrtc::VideoEncoder* encoder = factory_->CreateVideoEncoder(type);
|
| - if (encoder) {
|
| - non_simulcast_encoders_.push_back(encoder);
|
| - }
|
| - return encoder;
|
| - }
|
| -
|
| - const std::vector<VideoCodec>& codecs() const override {
|
| - return factory_->codecs();
|
| - }
|
| -
|
| - bool EncoderTypeHasInternalSource(
|
| - webrtc::VideoCodecType type) const override {
|
| - return factory_->EncoderTypeHasInternalSource(type);
|
| - }
|
| -
|
| - void DestroyVideoEncoder(webrtc::VideoEncoder* encoder) override {
|
| - // Check first to see if the encoder wasn't wrapped in a
|
| - // SimulcastEncoderAdapter. In that case, ask the factory to destroy it.
|
| - if (std::remove(non_simulcast_encoders_.begin(),
|
| - non_simulcast_encoders_.end(),
|
| - encoder) != non_simulcast_encoders_.end()) {
|
| - factory_->DestroyVideoEncoder(encoder);
|
| - return;
|
| - }
|
| -
|
| - // Otherwise, SimulcastEncoderAdapter can be deleted directly, and will call
|
| - // DestroyVideoEncoder on the factory for individual encoder instances.
|
| - delete encoder;
|
| - }
|
| -
|
| - private:
|
| - cricket::WebRtcVideoEncoderFactory* factory_;
|
| - // A list of encoders that were created without being wrapped in a
|
| - // SimulcastEncoderAdapter.
|
| - std::vector<webrtc::VideoEncoder*> non_simulcast_encoders_;
|
| -};
|
| -
|
| -bool CodecIsInternallySupported(const std::string& codec_name) {
|
| - if (CodecNamesEq(codec_name, kVp8CodecName)) {
|
| - return true;
|
| - }
|
| - if (CodecNamesEq(codec_name, kVp9CodecName)) {
|
| - return true;
|
| - }
|
| - if (CodecNamesEq(codec_name, kH264CodecName)) {
|
| - return webrtc::H264Encoder::IsSupported() &&
|
| - webrtc::H264Decoder::IsSupported();
|
| - }
|
| - return false;
|
| -}
|
| -
|
| -void AddDefaultFeedbackParams(VideoCodec* codec) {
|
| - codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamCcm, kRtcpFbCcmParamFir));
|
| - codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamNack, kParamValueEmpty));
|
| - codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamNack, kRtcpFbNackParamPli));
|
| - codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamRemb, kParamValueEmpty));
|
| - codec->AddFeedbackParam(
|
| - FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty));
|
| -}
|
| -
|
| -static VideoCodec MakeVideoCodecWithDefaultFeedbackParams(int payload_type,
|
| - const char* name) {
|
| - VideoCodec codec(payload_type, name, kDefaultVideoMaxWidth,
|
| - kDefaultVideoMaxHeight, kDefaultVideoMaxFramerate, 0);
|
| - AddDefaultFeedbackParams(&codec);
|
| - return codec;
|
| -}
|
| -
|
| -static std::string CodecVectorToString(const std::vector<VideoCodec>& codecs) {
|
| - std::stringstream out;
|
| - out << '{';
|
| - for (size_t i = 0; i < codecs.size(); ++i) {
|
| - out << codecs[i].ToString();
|
| - if (i != codecs.size() - 1) {
|
| - out << ", ";
|
| - }
|
| - }
|
| - out << '}';
|
| - return out.str();
|
| -}
|
| -
|
| -static bool ValidateCodecFormats(const std::vector<VideoCodec>& codecs) {
|
| - bool has_video = false;
|
| - for (size_t i = 0; i < codecs.size(); ++i) {
|
| - if (!codecs[i].ValidateCodecFormat()) {
|
| - return false;
|
| - }
|
| - if (codecs[i].GetCodecType() == VideoCodec::CODEC_VIDEO) {
|
| - has_video = true;
|
| - }
|
| - }
|
| - if (!has_video) {
|
| - LOG(LS_ERROR) << "Setting codecs without a video codec is invalid: "
|
| - << CodecVectorToString(codecs);
|
| - return false;
|
| - }
|
| - return true;
|
| -}
|
| -
|
| -static bool ValidateStreamParams(const StreamParams& sp) {
|
| - if (sp.ssrcs.empty()) {
|
| - LOG(LS_ERROR) << "No SSRCs in stream parameters: " << sp.ToString();
|
| - return false;
|
| - }
|
| -
|
| - std::vector<uint32_t> primary_ssrcs;
|
| - sp.GetPrimarySsrcs(&primary_ssrcs);
|
| - std::vector<uint32_t> rtx_ssrcs;
|
| - sp.GetFidSsrcs(primary_ssrcs, &rtx_ssrcs);
|
| - for (uint32_t rtx_ssrc : rtx_ssrcs) {
|
| - bool rtx_ssrc_present = false;
|
| - for (uint32_t sp_ssrc : sp.ssrcs) {
|
| - if (sp_ssrc == rtx_ssrc) {
|
| - rtx_ssrc_present = true;
|
| - break;
|
| - }
|
| - }
|
| - if (!rtx_ssrc_present) {
|
| - LOG(LS_ERROR) << "RTX SSRC '" << rtx_ssrc
|
| - << "' missing from StreamParams ssrcs: " << sp.ToString();
|
| - return false;
|
| - }
|
| - }
|
| - if (!rtx_ssrcs.empty() && primary_ssrcs.size() != rtx_ssrcs.size()) {
|
| - LOG(LS_ERROR)
|
| - << "RTX SSRCs exist, but don't cover all SSRCs (unsupported): "
|
| - << sp.ToString();
|
| - return false;
|
| - }
|
| -
|
| - return true;
|
| -}
|
| -
|
| -inline bool ContainsHeaderExtension(
|
| - const std::vector<webrtc::RtpExtension>& extensions,
|
| - const std::string& name) {
|
| - for (const auto& kv : extensions) {
|
| - if (kv.name == name) {
|
| - return true;
|
| - }
|
| - }
|
| - return false;
|
| -}
|
| -
|
| -// Merges two fec configs and logs an error if a conflict arises
|
| -// such that merging in different order would trigger a different output.
|
| -static void MergeFecConfig(const webrtc::FecConfig& other,
|
| - webrtc::FecConfig* output) {
|
| - if (other.ulpfec_payload_type != -1) {
|
| - if (output->ulpfec_payload_type != -1 &&
|
| - output->ulpfec_payload_type != other.ulpfec_payload_type) {
|
| - LOG(LS_WARNING) << "Conflict merging ulpfec_payload_type configs: "
|
| - << output->ulpfec_payload_type << " and "
|
| - << other.ulpfec_payload_type;
|
| - }
|
| - output->ulpfec_payload_type = other.ulpfec_payload_type;
|
| - }
|
| - if (other.red_payload_type != -1) {
|
| - if (output->red_payload_type != -1 &&
|
| - output->red_payload_type != other.red_payload_type) {
|
| - LOG(LS_WARNING) << "Conflict merging red_payload_type configs: "
|
| - << output->red_payload_type << " and "
|
| - << other.red_payload_type;
|
| - }
|
| - output->red_payload_type = other.red_payload_type;
|
| - }
|
| - if (other.red_rtx_payload_type != -1) {
|
| - if (output->red_rtx_payload_type != -1 &&
|
| - output->red_rtx_payload_type != other.red_rtx_payload_type) {
|
| - LOG(LS_WARNING) << "Conflict merging red_rtx_payload_type configs: "
|
| - << output->red_rtx_payload_type << " and "
|
| - << other.red_rtx_payload_type;
|
| - }
|
| - output->red_rtx_payload_type = other.red_rtx_payload_type;
|
| - }
|
| -}
|
| -
|
| -// Returns true if the given codec is disallowed from doing simulcast.
|
| -bool IsCodecBlacklistedForSimulcast(const std::string& codec_name) {
|
| - return CodecNamesEq(codec_name, kH264CodecName) ||
|
| - CodecNamesEq(codec_name, kVp9CodecName);
|
| -}
|
| -
|
| -// The selected thresholds for QVGA and VGA corresponded to a QP around 10.
|
| -// The change in QP declined above the selected bitrates.
|
| -static int GetMaxDefaultVideoBitrateKbps(int width, int height) {
|
| - if (width * height <= 320 * 240) {
|
| - return 600;
|
| - } else if (width * height <= 640 * 480) {
|
| - return 1700;
|
| - } else if (width * height <= 960 * 540) {
|
| - return 2000;
|
| - } else {
|
| - return 2500;
|
| - }
|
| -}
|
| -} // namespace
|
| -
|
| -// Constants defined in talk/media/webrtc/constants.h
|
| -// TODO(pbos): Move these to a separate constants.cc file.
|
| -const int kMinVideoBitrate = 30;
|
| -const int kStartVideoBitrate = 300;
|
| -
|
| -const int kVideoMtu = 1200;
|
| -const int kVideoRtpBufferSize = 65536;
|
| -
|
| -// This constant is really an on/off, lower-level configurable NACK history
|
| -// duration hasn't been implemented.
|
| -static const int kNackHistoryMs = 1000;
|
| -
|
| -static const int kDefaultQpMax = 56;
|
| -
|
| -static const int kDefaultRtcpReceiverReportSsrc = 1;
|
| -
|
| -std::vector<VideoCodec> DefaultVideoCodecList() {
|
| - std::vector<VideoCodec> codecs;
|
| - codecs.push_back(MakeVideoCodecWithDefaultFeedbackParams(kDefaultVp8PlType,
|
| - kVp8CodecName));
|
| - if (CodecIsInternallySupported(kVp9CodecName)) {
|
| - codecs.push_back(MakeVideoCodecWithDefaultFeedbackParams(kDefaultVp9PlType,
|
| - kVp9CodecName));
|
| - }
|
| - if (CodecIsInternallySupported(kH264CodecName)) {
|
| - codecs.push_back(MakeVideoCodecWithDefaultFeedbackParams(kDefaultH264PlType,
|
| - kH264CodecName));
|
| - }
|
| - codecs.push_back(
|
| - VideoCodec::CreateRtxCodec(kDefaultRtxVp8PlType, kDefaultVp8PlType));
|
| - if (CodecIsInternallySupported(kVp9CodecName)) {
|
| - codecs.push_back(
|
| - VideoCodec::CreateRtxCodec(kDefaultRtxVp9PlType, kDefaultVp9PlType));
|
| - }
|
| - codecs.push_back(VideoCodec(kDefaultRedPlType, kRedCodecName));
|
| - codecs.push_back(
|
| - VideoCodec::CreateRtxCodec(kDefaultRtxRedPlType, kDefaultRedPlType));
|
| - codecs.push_back(VideoCodec(kDefaultUlpfecType, kUlpfecCodecName));
|
| - return codecs;
|
| -}
|
| -
|
| -std::vector<webrtc::VideoStream>
|
| -WebRtcVideoChannel2::WebRtcVideoSendStream::CreateSimulcastVideoStreams(
|
| - const VideoCodec& codec,
|
| - const VideoOptions& options,
|
| - int max_bitrate_bps,
|
| - size_t num_streams) {
|
| - int max_qp = kDefaultQpMax;
|
| - codec.GetParam(kCodecParamMaxQuantization, &max_qp);
|
| -
|
| - return GetSimulcastConfig(
|
| - num_streams, codec.width, codec.height, max_bitrate_bps, max_qp,
|
| - codec.framerate != 0 ? codec.framerate : kDefaultVideoMaxFramerate);
|
| -}
|
| -
|
| -std::vector<webrtc::VideoStream>
|
| -WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoStreams(
|
| - const VideoCodec& codec,
|
| - const VideoOptions& options,
|
| - int max_bitrate_bps,
|
| - size_t num_streams) {
|
| - int codec_max_bitrate_kbps;
|
| - if (codec.GetParam(kCodecParamMaxBitrate, &codec_max_bitrate_kbps)) {
|
| - max_bitrate_bps = codec_max_bitrate_kbps * 1000;
|
| - }
|
| - if (num_streams != 1) {
|
| - return CreateSimulcastVideoStreams(codec, options, max_bitrate_bps,
|
| - num_streams);
|
| - }
|
| -
|
| - // For unset max bitrates set default bitrate for non-simulcast.
|
| - if (max_bitrate_bps <= 0) {
|
| - max_bitrate_bps =
|
| - GetMaxDefaultVideoBitrateKbps(codec.width, codec.height) * 1000;
|
| - }
|
| -
|
| - webrtc::VideoStream stream;
|
| - stream.width = codec.width;
|
| - stream.height = codec.height;
|
| - stream.max_framerate =
|
| - codec.framerate != 0 ? codec.framerate : kDefaultVideoMaxFramerate;
|
| -
|
| - stream.min_bitrate_bps = kMinVideoBitrate * 1000;
|
| - stream.target_bitrate_bps = stream.max_bitrate_bps = max_bitrate_bps;
|
| -
|
| - int max_qp = kDefaultQpMax;
|
| - codec.GetParam(kCodecParamMaxQuantization, &max_qp);
|
| - stream.max_qp = max_qp;
|
| - std::vector<webrtc::VideoStream> streams;
|
| - streams.push_back(stream);
|
| - return streams;
|
| -}
|
| -
|
| -void* WebRtcVideoChannel2::WebRtcVideoSendStream::ConfigureVideoEncoderSettings(
|
| - const VideoCodec& codec,
|
| - const VideoOptions& options,
|
| - bool is_screencast) {
|
| - // No automatic resizing when using simulcast or screencast.
|
| - bool automatic_resize =
|
| - !is_screencast && parameters_.config.rtp.ssrcs.size() == 1;
|
| - bool frame_dropping = !is_screencast;
|
| - bool denoising;
|
| - bool codec_default_denoising = false;
|
| - if (is_screencast) {
|
| - denoising = false;
|
| - } else {
|
| - // Use codec default if video_noise_reduction is unset.
|
| - codec_default_denoising = !options.video_noise_reduction;
|
| - denoising = options.video_noise_reduction.value_or(false);
|
| - }
|
| -
|
| - if (CodecNamesEq(codec.name, kH264CodecName)) {
|
| - encoder_settings_.h264 = webrtc::VideoEncoder::GetDefaultH264Settings();
|
| - encoder_settings_.h264.frameDroppingOn = frame_dropping;
|
| - return &encoder_settings_.h264;
|
| - }
|
| - if (CodecNamesEq(codec.name, kVp8CodecName)) {
|
| - encoder_settings_.vp8 = webrtc::VideoEncoder::GetDefaultVp8Settings();
|
| - encoder_settings_.vp8.automaticResizeOn = automatic_resize;
|
| - // VP8 denoising is enabled by default.
|
| - encoder_settings_.vp8.denoisingOn =
|
| - codec_default_denoising ? true : denoising;
|
| - encoder_settings_.vp8.frameDroppingOn = frame_dropping;
|
| - return &encoder_settings_.vp8;
|
| - }
|
| - if (CodecNamesEq(codec.name, kVp9CodecName)) {
|
| - encoder_settings_.vp9 = webrtc::VideoEncoder::GetDefaultVp9Settings();
|
| - // VP9 denoising is disabled by default.
|
| - encoder_settings_.vp9.denoisingOn =
|
| - codec_default_denoising ? false : denoising;
|
| - encoder_settings_.vp9.frameDroppingOn = frame_dropping;
|
| - return &encoder_settings_.vp9;
|
| - }
|
| - return NULL;
|
| -}
|
| -
|
| -DefaultUnsignalledSsrcHandler::DefaultUnsignalledSsrcHandler()
|
| - : default_recv_ssrc_(0), default_sink_(NULL) {}
|
| -
|
| -UnsignalledSsrcHandler::Action DefaultUnsignalledSsrcHandler::OnUnsignalledSsrc(
|
| - WebRtcVideoChannel2* channel,
|
| - uint32_t ssrc) {
|
| - if (default_recv_ssrc_ != 0) { // Already one default stream.
|
| - LOG(LS_WARNING) << "Unknown SSRC, but default receive stream already set.";
|
| - return kDropPacket;
|
| - }
|
| -
|
| - StreamParams sp;
|
| - sp.ssrcs.push_back(ssrc);
|
| - LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << ".";
|
| - if (!channel->AddRecvStream(sp, true)) {
|
| - LOG(LS_WARNING) << "Could not create default receive stream.";
|
| - }
|
| -
|
| - channel->SetSink(ssrc, default_sink_);
|
| - default_recv_ssrc_ = ssrc;
|
| - return kDeliverPacket;
|
| -}
|
| -
|
| -rtc::VideoSinkInterface<VideoFrame>*
|
| -DefaultUnsignalledSsrcHandler::GetDefaultSink() const {
|
| - return default_sink_;
|
| -}
|
| -
|
| -void DefaultUnsignalledSsrcHandler::SetDefaultSink(
|
| - VideoMediaChannel* channel,
|
| - rtc::VideoSinkInterface<VideoFrame>* sink) {
|
| - default_sink_ = sink;
|
| - if (default_recv_ssrc_ != 0) {
|
| - channel->SetSink(default_recv_ssrc_, default_sink_);
|
| - }
|
| -}
|
| -
|
| -WebRtcVideoEngine2::WebRtcVideoEngine2()
|
| - : initialized_(false),
|
| - external_decoder_factory_(NULL),
|
| - external_encoder_factory_(NULL) {
|
| - LOG(LS_INFO) << "WebRtcVideoEngine2::WebRtcVideoEngine2()";
|
| - video_codecs_ = GetSupportedCodecs();
|
| -}
|
| -
|
| -WebRtcVideoEngine2::~WebRtcVideoEngine2() {
|
| - LOG(LS_INFO) << "WebRtcVideoEngine2::~WebRtcVideoEngine2";
|
| -}
|
| -
|
| -void WebRtcVideoEngine2::Init() {
|
| - LOG(LS_INFO) << "WebRtcVideoEngine2::Init";
|
| - initialized_ = true;
|
| -}
|
| -
|
| -WebRtcVideoChannel2* WebRtcVideoEngine2::CreateChannel(
|
| - webrtc::Call* call,
|
| - const VideoOptions& options) {
|
| - RTC_DCHECK(initialized_);
|
| - LOG(LS_INFO) << "CreateChannel. Options: " << options.ToString();
|
| - return new WebRtcVideoChannel2(call, options, video_codecs_,
|
| - external_encoder_factory_, external_decoder_factory_);
|
| -}
|
| -
|
| -const std::vector<VideoCodec>& WebRtcVideoEngine2::codecs() const {
|
| - return video_codecs_;
|
| -}
|
| -
|
| -RtpCapabilities WebRtcVideoEngine2::GetCapabilities() const {
|
| - RtpCapabilities capabilities;
|
| - capabilities.header_extensions.push_back(
|
| - RtpHeaderExtension(kRtpTimestampOffsetHeaderExtension,
|
| - kRtpTimestampOffsetHeaderExtensionDefaultId));
|
| - capabilities.header_extensions.push_back(
|
| - RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
|
| - kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
|
| - capabilities.header_extensions.push_back(
|
| - RtpHeaderExtension(kRtpVideoRotationHeaderExtension,
|
| - kRtpVideoRotationHeaderExtensionDefaultId));
|
| - if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") {
|
| - capabilities.header_extensions.push_back(RtpHeaderExtension(
|
| - kRtpTransportSequenceNumberHeaderExtension,
|
| - kRtpTransportSequenceNumberHeaderExtensionDefaultId));
|
| - }
|
| - return capabilities;
|
| -}
|
| -
|
| -void WebRtcVideoEngine2::SetExternalDecoderFactory(
|
| - WebRtcVideoDecoderFactory* decoder_factory) {
|
| - RTC_DCHECK(!initialized_);
|
| - external_decoder_factory_ = decoder_factory;
|
| -}
|
| -
|
| -void WebRtcVideoEngine2::SetExternalEncoderFactory(
|
| - WebRtcVideoEncoderFactory* encoder_factory) {
|
| - RTC_DCHECK(!initialized_);
|
| - if (external_encoder_factory_ == encoder_factory)
|
| - return;
|
| -
|
| - // No matter what happens we shouldn't hold on to a stale
|
| - // WebRtcSimulcastEncoderFactory.
|
| - simulcast_encoder_factory_.reset();
|
| -
|
| - if (encoder_factory &&
|
| - WebRtcSimulcastEncoderFactory::UseSimulcastEncoderFactory(
|
| - encoder_factory->codecs())) {
|
| - simulcast_encoder_factory_.reset(
|
| - new WebRtcSimulcastEncoderFactory(encoder_factory));
|
| - encoder_factory = simulcast_encoder_factory_.get();
|
| - }
|
| - external_encoder_factory_ = encoder_factory;
|
| -
|
| - video_codecs_ = GetSupportedCodecs();
|
| -}
|
| -
|
| -std::vector<VideoCodec> WebRtcVideoEngine2::GetSupportedCodecs() const {
|
| - std::vector<VideoCodec> supported_codecs = DefaultVideoCodecList();
|
| -
|
| - if (external_encoder_factory_ == NULL) {
|
| - return supported_codecs;
|
| - }
|
| -
|
| - const std::vector<WebRtcVideoEncoderFactory::VideoCodec>& codecs =
|
| - external_encoder_factory_->codecs();
|
| - for (size_t i = 0; i < codecs.size(); ++i) {
|
| - // Don't add internally-supported codecs twice.
|
| - if (CodecIsInternallySupported(codecs[i].name)) {
|
| - continue;
|
| - }
|
| -
|
| - // External video encoders are given payloads 120-127. This also means that
|
| - // we only support up to 8 external payload types.
|
| - const int kExternalVideoPayloadTypeBase = 120;
|
| - size_t payload_type = kExternalVideoPayloadTypeBase + i;
|
| - RTC_DCHECK(payload_type < 128);
|
| - VideoCodec codec(static_cast<int>(payload_type),
|
| - codecs[i].name,
|
| - codecs[i].max_width,
|
| - codecs[i].max_height,
|
| - codecs[i].max_fps,
|
| - 0);
|
| -
|
| - AddDefaultFeedbackParams(&codec);
|
| - supported_codecs.push_back(codec);
|
| - }
|
| - return supported_codecs;
|
| -}
|
| -
|
| -WebRtcVideoChannel2::WebRtcVideoChannel2(
|
| - webrtc::Call* call,
|
| - const VideoOptions& options,
|
| - const std::vector<VideoCodec>& recv_codecs,
|
| - WebRtcVideoEncoderFactory* external_encoder_factory,
|
| - WebRtcVideoDecoderFactory* external_decoder_factory)
|
| - : call_(call),
|
| - unsignalled_ssrc_handler_(&default_unsignalled_ssrc_handler_),
|
| - external_encoder_factory_(external_encoder_factory),
|
| - external_decoder_factory_(external_decoder_factory) {
|
| - RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
| - SetDefaultOptions();
|
| - options_.SetAll(options);
|
| - if (options_.cpu_overuse_detection)
|
| - signal_cpu_adaptation_ = *options_.cpu_overuse_detection;
|
| - rtcp_receiver_report_ssrc_ = kDefaultRtcpReceiverReportSsrc;
|
| - sending_ = false;
|
| - default_send_ssrc_ = 0;
|
| - RTC_DCHECK(ValidateCodecFormats(recv_codecs));
|
| - recv_codecs_ = FilterSupportedCodecs(MapCodecs(recv_codecs));
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::SetDefaultOptions() {
|
| - options_.cpu_overuse_detection = rtc::Optional<bool>(true);
|
| - options_.dscp = rtc::Optional<bool>(false);
|
| - options_.suspend_below_min_bitrate = rtc::Optional<bool>(false);
|
| - options_.screencast_min_bitrate_kbps = rtc::Optional<int>(0);
|
| -}
|
| -
|
| -WebRtcVideoChannel2::~WebRtcVideoChannel2() {
|
| - for (auto& kv : send_streams_)
|
| - delete kv.second;
|
| - for (auto& kv : receive_streams_)
|
| - delete kv.second;
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::CodecIsExternallySupported(
|
| - const std::string& name) const {
|
| - if (external_encoder_factory_ == NULL) {
|
| - return false;
|
| - }
|
| -
|
| - const std::vector<WebRtcVideoEncoderFactory::VideoCodec> external_codecs =
|
| - external_encoder_factory_->codecs();
|
| - for (size_t c = 0; c < external_codecs.size(); ++c) {
|
| - if (CodecNamesEq(name, external_codecs[c].name)) {
|
| - return true;
|
| - }
|
| - }
|
| - return false;
|
| -}
|
| -
|
| -std::vector<WebRtcVideoChannel2::VideoCodecSettings>
|
| -WebRtcVideoChannel2::FilterSupportedCodecs(
|
| - const std::vector<WebRtcVideoChannel2::VideoCodecSettings>& mapped_codecs)
|
| - const {
|
| - std::vector<VideoCodecSettings> supported_codecs;
|
| - for (size_t i = 0; i < mapped_codecs.size(); ++i) {
|
| - const VideoCodecSettings& codec = mapped_codecs[i];
|
| - if (CodecIsInternallySupported(codec.codec.name) ||
|
| - CodecIsExternallySupported(codec.codec.name)) {
|
| - supported_codecs.push_back(codec);
|
| - }
|
| - }
|
| - return supported_codecs;
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::ReceiveCodecsHaveChanged(
|
| - std::vector<VideoCodecSettings> before,
|
| - std::vector<VideoCodecSettings> after) {
|
| - if (before.size() != after.size()) {
|
| - return true;
|
| - }
|
| - // The receive codec order doesn't matter, so we sort the codecs before
|
| - // comparing. This is necessary because currently the
|
| - // only way to change the send codec is to munge SDP, which causes
|
| - // the receive codec list to change order, which causes the streams
|
| - // to be recreates which causes a "blink" of black video. In order
|
| - // to support munging the SDP in this way without recreating receive
|
| - // streams, we ignore the order of the received codecs so that
|
| - // changing the order doesn't cause this "blink".
|
| - auto comparison =
|
| - [](const VideoCodecSettings& codec1, const VideoCodecSettings& codec2) {
|
| - return codec1.codec.id > codec2.codec.id;
|
| - };
|
| - std::sort(before.begin(), before.end(), comparison);
|
| - std::sort(after.begin(), after.end(), comparison);
|
| - for (size_t i = 0; i < before.size(); ++i) {
|
| - // For the same reason that we sort the codecs, we also ignore the
|
| - // preference. We don't want a preference change on the receive
|
| - // side to cause recreation of the stream.
|
| - before[i].codec.preference = 0;
|
| - after[i].codec.preference = 0;
|
| - if (before[i] != after[i]) {
|
| - return true;
|
| - }
|
| - }
|
| - return false;
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::GetChangedSendParameters(
|
| - const VideoSendParameters& params,
|
| - ChangedSendParameters* changed_params) const {
|
| - if (!ValidateCodecFormats(params.codecs) ||
|
| - !ValidateRtpExtensions(params.extensions)) {
|
| - return false;
|
| - }
|
| -
|
| - // Handle send codec.
|
| - const std::vector<VideoCodecSettings> supported_codecs =
|
| - FilterSupportedCodecs(MapCodecs(params.codecs));
|
| -
|
| - if (supported_codecs.empty()) {
|
| - LOG(LS_ERROR) << "No video codecs supported.";
|
| - return false;
|
| - }
|
| -
|
| - if (!send_codec_ || supported_codecs.front() != *send_codec_) {
|
| - changed_params->codec =
|
| - rtc::Optional<VideoCodecSettings>(supported_codecs.front());
|
| - }
|
| -
|
| - // Handle RTP header extensions.
|
| - std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
|
| - params.extensions, webrtc::RtpExtension::IsSupportedForVideo, true);
|
| - if (send_rtp_extensions_ != filtered_extensions) {
|
| - changed_params->rtp_header_extensions =
|
| - rtc::Optional<std::vector<webrtc::RtpExtension>>(filtered_extensions);
|
| - }
|
| -
|
| - // Handle max bitrate.
|
| - if (params.max_bandwidth_bps != bitrate_config_.max_bitrate_bps &&
|
| - params.max_bandwidth_bps >= 0) {
|
| - // 0 uncaps max bitrate (-1).
|
| - changed_params->max_bandwidth_bps = rtc::Optional<int>(
|
| - params.max_bandwidth_bps == 0 ? -1 : params.max_bandwidth_bps);
|
| - }
|
| -
|
| - // Handle options.
|
| - // TODO(pbos): Require VideoSendParameters to contain a full set of options
|
| - // and check if params.options != options_ instead of applying a delta.
|
| - VideoOptions new_options = options_;
|
| - new_options.SetAll(params.options);
|
| - if (!(new_options == options_)) {
|
| - changed_params->options = rtc::Optional<VideoOptions>(new_options);
|
| - }
|
| -
|
| - // Handle RTCP mode.
|
| - if (params.rtcp.reduced_size != send_params_.rtcp.reduced_size) {
|
| - changed_params->rtcp_mode = rtc::Optional<webrtc::RtcpMode>(
|
| - params.rtcp.reduced_size ? webrtc::RtcpMode::kReducedSize
|
| - : webrtc::RtcpMode::kCompound);
|
| - }
|
| -
|
| - return true;
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::SetSendParameters(const VideoSendParameters& params) {
|
| - TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetSendParameters");
|
| - LOG(LS_INFO) << "SetSendParameters: " << params.ToString();
|
| - ChangedSendParameters changed_params;
|
| - if (!GetChangedSendParameters(params, &changed_params)) {
|
| - return false;
|
| - }
|
| -
|
| - bool bitrate_config_changed = false;
|
| -
|
| - if (changed_params.codec) {
|
| - const VideoCodecSettings& codec_settings = *changed_params.codec;
|
| - send_codec_ = rtc::Optional<VideoCodecSettings>(codec_settings);
|
| -
|
| - LOG(LS_INFO) << "Using codec: " << codec_settings.codec.ToString();
|
| - // TODO(holmer): Changing the codec parameters shouldn't necessarily mean
|
| - // that we change the min/max of bandwidth estimation. Reevaluate this.
|
| - bitrate_config_ = GetBitrateConfigForCodec(codec_settings.codec);
|
| - bitrate_config_changed = true;
|
| - }
|
| -
|
| - if (changed_params.rtp_header_extensions) {
|
| - send_rtp_extensions_ = *changed_params.rtp_header_extensions;
|
| - }
|
| -
|
| - if (changed_params.max_bandwidth_bps) {
|
| - // TODO(pbos): Figure out whether b=AS means max bitrate for this
|
| - // WebRtcVideoChannel2 (in which case we're good), or per sender (SSRC), in
|
| - // which case this should not set a Call::BitrateConfig but rather
|
| - // reconfigure all senders.
|
| - int max_bitrate_bps = *changed_params.max_bandwidth_bps;
|
| - bitrate_config_.start_bitrate_bps = -1;
|
| - bitrate_config_.max_bitrate_bps = max_bitrate_bps;
|
| - if (max_bitrate_bps > 0 &&
|
| - bitrate_config_.min_bitrate_bps > max_bitrate_bps) {
|
| - bitrate_config_.min_bitrate_bps = max_bitrate_bps;
|
| - }
|
| - bitrate_config_changed = true;
|
| - }
|
| -
|
| - if (bitrate_config_changed) {
|
| - call_->SetBitrateConfig(bitrate_config_);
|
| - }
|
| -
|
| - if (changed_params.options) {
|
| - options_.SetAll(*changed_params.options);
|
| - {
|
| - rtc::CritScope lock(&capturer_crit_);
|
| - if (options_.cpu_overuse_detection) {
|
| - signal_cpu_adaptation_ = *options_.cpu_overuse_detection;
|
| - }
|
| - }
|
| - rtc::DiffServCodePoint dscp =
|
| - options_.dscp.value_or(false) ? rtc::DSCP_AF41 : rtc::DSCP_DEFAULT;
|
| - MediaChannel::SetDscp(dscp);
|
| - }
|
| -
|
| - {
|
| - rtc::CritScope stream_lock(&stream_crit_);
|
| - for (auto& kv : send_streams_) {
|
| - kv.second->SetSendParameters(changed_params);
|
| - }
|
| - if (changed_params.codec) {
|
| - // Update receive feedback parameters from new codec.
|
| - LOG(LS_INFO)
|
| - << "SetFeedbackOptions on all the receive streams because the send "
|
| - "codec has changed.";
|
| - for (auto& kv : receive_streams_) {
|
| - RTC_DCHECK(kv.second != nullptr);
|
| - kv.second->SetFeedbackParameters(HasNack(send_codec_->codec),
|
| - HasRemb(send_codec_->codec),
|
| - HasTransportCc(send_codec_->codec));
|
| - }
|
| - }
|
| - }
|
| - send_params_ = params;
|
| - return true;
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::GetChangedRecvParameters(
|
| - const VideoRecvParameters& params,
|
| - ChangedRecvParameters* changed_params) const {
|
| - if (!ValidateCodecFormats(params.codecs) ||
|
| - !ValidateRtpExtensions(params.extensions)) {
|
| - return false;
|
| - }
|
| -
|
| - // Handle receive codecs.
|
| - const std::vector<VideoCodecSettings> mapped_codecs =
|
| - MapCodecs(params.codecs);
|
| - if (mapped_codecs.empty()) {
|
| - LOG(LS_ERROR) << "SetRecvParameters called without any video codecs.";
|
| - return false;
|
| - }
|
| -
|
| - std::vector<VideoCodecSettings> supported_codecs =
|
| - FilterSupportedCodecs(mapped_codecs);
|
| -
|
| - if (mapped_codecs.size() != supported_codecs.size()) {
|
| - LOG(LS_ERROR) << "SetRecvParameters called with unsupported video codecs.";
|
| - return false;
|
| - }
|
| -
|
| - if (ReceiveCodecsHaveChanged(recv_codecs_, supported_codecs)) {
|
| - changed_params->codec_settings =
|
| - rtc::Optional<std::vector<VideoCodecSettings>>(supported_codecs);
|
| - }
|
| -
|
| - // Handle RTP header extensions.
|
| - std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
|
| - params.extensions, webrtc::RtpExtension::IsSupportedForVideo, false);
|
| - if (filtered_extensions != recv_rtp_extensions_) {
|
| - changed_params->rtp_header_extensions =
|
| - rtc::Optional<std::vector<webrtc::RtpExtension>>(filtered_extensions);
|
| - }
|
| -
|
| - // Handle RTCP mode.
|
| - if (params.rtcp.reduced_size != recv_params_.rtcp.reduced_size) {
|
| - changed_params->rtcp_mode = rtc::Optional<webrtc::RtcpMode>(
|
| - params.rtcp.reduced_size ? webrtc::RtcpMode::kReducedSize
|
| - : webrtc::RtcpMode::kCompound);
|
| - }
|
| -
|
| - return true;
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::SetRecvParameters(const VideoRecvParameters& params) {
|
| - TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRecvParameters");
|
| - LOG(LS_INFO) << "SetRecvParameters: " << params.ToString();
|
| - ChangedRecvParameters changed_params;
|
| - if (!GetChangedRecvParameters(params, &changed_params)) {
|
| - return false;
|
| - }
|
| - if (changed_params.rtp_header_extensions) {
|
| - recv_rtp_extensions_ = *changed_params.rtp_header_extensions;
|
| - }
|
| - if (changed_params.codec_settings) {
|
| - LOG(LS_INFO) << "Changing recv codecs from "
|
| - << CodecSettingsVectorToString(recv_codecs_) << " to "
|
| - << CodecSettingsVectorToString(*changed_params.codec_settings);
|
| - recv_codecs_ = *changed_params.codec_settings;
|
| - }
|
| -
|
| - {
|
| - rtc::CritScope stream_lock(&stream_crit_);
|
| - for (auto& kv : receive_streams_) {
|
| - kv.second->SetRecvParameters(changed_params);
|
| - }
|
| - }
|
| - recv_params_ = params;
|
| - return true;
|
| -}
|
| -
|
| -std::string WebRtcVideoChannel2::CodecSettingsVectorToString(
|
| - const std::vector<VideoCodecSettings>& codecs) {
|
| - std::stringstream out;
|
| - out << '{';
|
| - for (size_t i = 0; i < codecs.size(); ++i) {
|
| - out << codecs[i].codec.ToString();
|
| - if (i != codecs.size() - 1) {
|
| - out << ", ";
|
| - }
|
| - }
|
| - out << '}';
|
| - return out.str();
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::GetSendCodec(VideoCodec* codec) {
|
| - if (!send_codec_) {
|
| - LOG(LS_VERBOSE) << "GetSendCodec: No send codec set.";
|
| - return false;
|
| - }
|
| - *codec = send_codec_->codec;
|
| - return true;
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::SetSend(bool send) {
|
| - LOG(LS_VERBOSE) << "SetSend: " << (send ? "true" : "false");
|
| - if (send && !send_codec_) {
|
| - LOG(LS_ERROR) << "SetSend(true) called before setting codec.";
|
| - return false;
|
| - }
|
| - if (send) {
|
| - StartAllSendStreams();
|
| - } else {
|
| - StopAllSendStreams();
|
| - }
|
| - sending_ = send;
|
| - return true;
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::SetVideoSend(uint32_t ssrc, bool enable,
|
| - const VideoOptions* options) {
|
| - TRACE_EVENT0("webrtc", "SetVideoSend");
|
| - LOG(LS_INFO) << "SetVideoSend (ssrc= " << ssrc << ", enable = " << enable
|
| - << "options: " << (options ? options->ToString() : "nullptr")
|
| - << ").";
|
| -
|
| - // TODO(solenberg): The state change should be fully rolled back if any one of
|
| - // these calls fail.
|
| - if (!MuteStream(ssrc, !enable)) {
|
| - return false;
|
| - }
|
| - if (enable && options) {
|
| - VideoSendParameters new_params = send_params_;
|
| - new_params.options.SetAll(*options);
|
| - SetSendParameters(send_params_);
|
| - }
|
| - return true;
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::ValidateSendSsrcAvailability(
|
| - const StreamParams& sp) const {
|
| - for (uint32_t ssrc: sp.ssrcs) {
|
| - if (send_ssrcs_.find(ssrc) != send_ssrcs_.end()) {
|
| - LOG(LS_ERROR) << "Send stream with SSRC '" << ssrc << "' already exists.";
|
| - return false;
|
| - }
|
| - }
|
| - return true;
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::ValidateReceiveSsrcAvailability(
|
| - const StreamParams& sp) const {
|
| - for (uint32_t ssrc: sp.ssrcs) {
|
| - if (receive_ssrcs_.find(ssrc) != receive_ssrcs_.end()) {
|
| - LOG(LS_ERROR) << "Receive stream with SSRC '" << ssrc
|
| - << "' already exists.";
|
| - return false;
|
| - }
|
| - }
|
| - return true;
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::AddSendStream(const StreamParams& sp) {
|
| - LOG(LS_INFO) << "AddSendStream: " << sp.ToString();
|
| - if (!ValidateStreamParams(sp))
|
| - return false;
|
| -
|
| - rtc::CritScope stream_lock(&stream_crit_);
|
| -
|
| - if (!ValidateSendSsrcAvailability(sp))
|
| - return false;
|
| -
|
| - for (uint32_t used_ssrc : sp.ssrcs)
|
| - send_ssrcs_.insert(used_ssrc);
|
| -
|
| - webrtc::VideoSendStream::Config config(this);
|
| - config.overuse_callback = this;
|
| -
|
| - WebRtcVideoSendStream* stream = new WebRtcVideoSendStream(
|
| - call_, sp, config, external_encoder_factory_, options_,
|
| - bitrate_config_.max_bitrate_bps, send_codec_, send_rtp_extensions_,
|
| - send_params_);
|
| -
|
| - uint32_t ssrc = sp.first_ssrc();
|
| - RTC_DCHECK(ssrc != 0);
|
| - send_streams_[ssrc] = stream;
|
| -
|
| - if (rtcp_receiver_report_ssrc_ == kDefaultRtcpReceiverReportSsrc) {
|
| - rtcp_receiver_report_ssrc_ = ssrc;
|
| - LOG(LS_INFO) << "SetLocalSsrc on all the receive streams because we added "
|
| - "a send stream.";
|
| - for (auto& kv : receive_streams_)
|
| - kv.second->SetLocalSsrc(ssrc);
|
| - }
|
| - if (default_send_ssrc_ == 0) {
|
| - default_send_ssrc_ = ssrc;
|
| - }
|
| - if (sending_) {
|
| - stream->Start();
|
| - }
|
| -
|
| - return true;
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::RemoveSendStream(uint32_t ssrc) {
|
| - LOG(LS_INFO) << "RemoveSendStream: " << ssrc;
|
| -
|
| - if (ssrc == 0) {
|
| - if (default_send_ssrc_ == 0) {
|
| - LOG(LS_ERROR) << "No default send stream active.";
|
| - return false;
|
| - }
|
| -
|
| - LOG(LS_VERBOSE) << "Removing default stream: " << default_send_ssrc_;
|
| - ssrc = default_send_ssrc_;
|
| - }
|
| -
|
| - WebRtcVideoSendStream* removed_stream;
|
| - {
|
| - rtc::CritScope stream_lock(&stream_crit_);
|
| - std::map<uint32_t, WebRtcVideoSendStream*>::iterator it =
|
| - send_streams_.find(ssrc);
|
| - if (it == send_streams_.end()) {
|
| - return false;
|
| - }
|
| -
|
| - for (uint32_t old_ssrc : it->second->GetSsrcs())
|
| - send_ssrcs_.erase(old_ssrc);
|
| -
|
| - removed_stream = it->second;
|
| - send_streams_.erase(it);
|
| -
|
| - // Switch receiver report SSRCs, the one in use is no longer valid.
|
| - if (rtcp_receiver_report_ssrc_ == ssrc) {
|
| - rtcp_receiver_report_ssrc_ = send_streams_.empty()
|
| - ? kDefaultRtcpReceiverReportSsrc
|
| - : send_streams_.begin()->first;
|
| - LOG(LS_INFO) << "SetLocalSsrc on all the receive streams because the "
|
| - "previous local SSRC was removed.";
|
| -
|
| - for (auto& kv : receive_streams_) {
|
| - kv.second->SetLocalSsrc(rtcp_receiver_report_ssrc_);
|
| - }
|
| - }
|
| - }
|
| -
|
| - delete removed_stream;
|
| -
|
| - if (ssrc == default_send_ssrc_) {
|
| - default_send_ssrc_ = 0;
|
| - }
|
| -
|
| - return true;
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::DeleteReceiveStream(
|
| - WebRtcVideoChannel2::WebRtcVideoReceiveStream* stream) {
|
| - for (uint32_t old_ssrc : stream->GetSsrcs())
|
| - receive_ssrcs_.erase(old_ssrc);
|
| - delete stream;
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::AddRecvStream(const StreamParams& sp) {
|
| - return AddRecvStream(sp, false);
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::AddRecvStream(const StreamParams& sp,
|
| - bool default_stream) {
|
| - RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
| -
|
| - LOG(LS_INFO) << "AddRecvStream" << (default_stream ? " (default stream)" : "")
|
| - << ": " << sp.ToString();
|
| - if (!ValidateStreamParams(sp))
|
| - return false;
|
| -
|
| - uint32_t ssrc = sp.first_ssrc();
|
| - RTC_DCHECK(ssrc != 0); // TODO(pbos): Is this ever valid?
|
| -
|
| - rtc::CritScope stream_lock(&stream_crit_);
|
| - // Remove running stream if this was a default stream.
|
| - auto prev_stream = receive_streams_.find(ssrc);
|
| - if (prev_stream != receive_streams_.end()) {
|
| - if (default_stream || !prev_stream->second->IsDefaultStream()) {
|
| - LOG(LS_ERROR) << "Receive stream for SSRC '" << ssrc
|
| - << "' already exists.";
|
| - return false;
|
| - }
|
| - DeleteReceiveStream(prev_stream->second);
|
| - receive_streams_.erase(prev_stream);
|
| - }
|
| -
|
| - if (!ValidateReceiveSsrcAvailability(sp))
|
| - return false;
|
| -
|
| - for (uint32_t used_ssrc : sp.ssrcs)
|
| - receive_ssrcs_.insert(used_ssrc);
|
| -
|
| - webrtc::VideoReceiveStream::Config config(this);
|
| - ConfigureReceiverRtp(&config, sp);
|
| -
|
| - // Set up A/V sync group based on sync label.
|
| - config.sync_group = sp.sync_label;
|
| -
|
| - config.rtp.remb = send_codec_ ? HasRemb(send_codec_->codec) : false;
|
| - config.rtp.transport_cc =
|
| - send_codec_ ? HasTransportCc(send_codec_->codec) : false;
|
| -
|
| - receive_streams_[ssrc] = new WebRtcVideoReceiveStream(
|
| - call_, sp, config, external_decoder_factory_, default_stream,
|
| - recv_codecs_, options_.disable_prerenderer_smoothing.value_or(false));
|
| -
|
| - return true;
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::ConfigureReceiverRtp(
|
| - webrtc::VideoReceiveStream::Config* config,
|
| - const StreamParams& sp) const {
|
| - uint32_t ssrc = sp.first_ssrc();
|
| -
|
| - config->rtp.remote_ssrc = ssrc;
|
| - config->rtp.local_ssrc = rtcp_receiver_report_ssrc_;
|
| -
|
| - config->rtp.extensions = recv_rtp_extensions_;
|
| - config->rtp.rtcp_mode = recv_params_.rtcp.reduced_size
|
| - ? webrtc::RtcpMode::kReducedSize
|
| - : webrtc::RtcpMode::kCompound;
|
| -
|
| - // TODO(pbos): This protection is against setting the same local ssrc as
|
| - // remote which is not permitted by the lower-level API. RTCP requires a
|
| - // corresponding sender SSRC. Figure out what to do when we don't have
|
| - // (receive-only) or know a good local SSRC.
|
| - if (config->rtp.remote_ssrc == config->rtp.local_ssrc) {
|
| - if (config->rtp.local_ssrc != kDefaultRtcpReceiverReportSsrc) {
|
| - config->rtp.local_ssrc = kDefaultRtcpReceiverReportSsrc;
|
| - } else {
|
| - config->rtp.local_ssrc = kDefaultRtcpReceiverReportSsrc + 1;
|
| - }
|
| - }
|
| -
|
| - for (size_t i = 0; i < recv_codecs_.size(); ++i) {
|
| - MergeFecConfig(recv_codecs_[i].fec, &config->rtp.fec);
|
| - }
|
| -
|
| - for (size_t i = 0; i < recv_codecs_.size(); ++i) {
|
| - uint32_t rtx_ssrc;
|
| - if (recv_codecs_[i].rtx_payload_type != -1 &&
|
| - sp.GetFidSsrc(ssrc, &rtx_ssrc)) {
|
| - webrtc::VideoReceiveStream::Config::Rtp::Rtx& rtx =
|
| - config->rtp.rtx[recv_codecs_[i].codec.id];
|
| - rtx.ssrc = rtx_ssrc;
|
| - rtx.payload_type = recv_codecs_[i].rtx_payload_type;
|
| - }
|
| - }
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::RemoveRecvStream(uint32_t ssrc) {
|
| - LOG(LS_INFO) << "RemoveRecvStream: " << ssrc;
|
| - if (ssrc == 0) {
|
| - LOG(LS_ERROR) << "RemoveRecvStream with 0 ssrc is not supported.";
|
| - return false;
|
| - }
|
| -
|
| - rtc::CritScope stream_lock(&stream_crit_);
|
| - std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator stream =
|
| - receive_streams_.find(ssrc);
|
| - if (stream == receive_streams_.end()) {
|
| - LOG(LS_ERROR) << "Stream not found for ssrc: " << ssrc;
|
| - return false;
|
| - }
|
| - DeleteReceiveStream(stream->second);
|
| - receive_streams_.erase(stream);
|
| -
|
| - return true;
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::SetSink(uint32_t ssrc,
|
| - rtc::VideoSinkInterface<VideoFrame>* sink) {
|
| - LOG(LS_INFO) << "SetSink: ssrc:" << ssrc << " " << (sink ? "(ptr)" : "NULL");
|
| - if (ssrc == 0) {
|
| - default_unsignalled_ssrc_handler_.SetDefaultSink(this, sink);
|
| - return true;
|
| - }
|
| -
|
| - rtc::CritScope stream_lock(&stream_crit_);
|
| - std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator it =
|
| - receive_streams_.find(ssrc);
|
| - if (it == receive_streams_.end()) {
|
| - return false;
|
| - }
|
| -
|
| - it->second->SetSink(sink);
|
| - return true;
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::GetStats(VideoMediaInfo* info) {
|
| - info->Clear();
|
| - FillSenderStats(info);
|
| - FillReceiverStats(info);
|
| - webrtc::Call::Stats stats = call_->GetStats();
|
| - FillBandwidthEstimationStats(stats, info);
|
| - if (stats.rtt_ms != -1) {
|
| - for (size_t i = 0; i < info->senders.size(); ++i) {
|
| - info->senders[i].rtt_ms = stats.rtt_ms;
|
| - }
|
| - }
|
| - return true;
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::FillSenderStats(VideoMediaInfo* video_media_info) {
|
| - rtc::CritScope stream_lock(&stream_crit_);
|
| - for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator it =
|
| - send_streams_.begin();
|
| - it != send_streams_.end(); ++it) {
|
| - video_media_info->senders.push_back(it->second->GetVideoSenderInfo());
|
| - }
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::FillReceiverStats(VideoMediaInfo* video_media_info) {
|
| - rtc::CritScope stream_lock(&stream_crit_);
|
| - for (std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator it =
|
| - receive_streams_.begin();
|
| - it != receive_streams_.end(); ++it) {
|
| - video_media_info->receivers.push_back(it->second->GetVideoReceiverInfo());
|
| - }
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::FillBandwidthEstimationStats(
|
| - const webrtc::Call::Stats& stats,
|
| - VideoMediaInfo* video_media_info) {
|
| - BandwidthEstimationInfo bwe_info;
|
| - bwe_info.available_send_bandwidth = stats.send_bandwidth_bps;
|
| - bwe_info.available_recv_bandwidth = stats.recv_bandwidth_bps;
|
| - bwe_info.bucket_delay = stats.pacer_delay_ms;
|
| -
|
| - // Get send stream bitrate stats.
|
| - rtc::CritScope stream_lock(&stream_crit_);
|
| - for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator stream =
|
| - send_streams_.begin();
|
| - stream != send_streams_.end(); ++stream) {
|
| - stream->second->FillBandwidthEstimationInfo(&bwe_info);
|
| - }
|
| - video_media_info->bw_estimations.push_back(bwe_info);
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::SetCapturer(uint32_t ssrc, VideoCapturer* capturer) {
|
| - LOG(LS_INFO) << "SetCapturer: " << ssrc << " -> "
|
| - << (capturer != NULL ? "(capturer)" : "NULL");
|
| - RTC_DCHECK(ssrc != 0);
|
| - {
|
| - rtc::CritScope stream_lock(&stream_crit_);
|
| - if (send_streams_.find(ssrc) == send_streams_.end()) {
|
| - LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc;
|
| - return false;
|
| - }
|
| - if (!send_streams_[ssrc]->SetCapturer(capturer)) {
|
| - return false;
|
| - }
|
| - }
|
| -
|
| - if (capturer) {
|
| - capturer->SetApplyRotation(!ContainsHeaderExtension(
|
| - send_rtp_extensions_, kRtpVideoRotationHeaderExtension));
|
| - }
|
| - {
|
| - rtc::CritScope lock(&capturer_crit_);
|
| - capturers_[ssrc] = capturer;
|
| - }
|
| - return true;
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::OnPacketReceived(
|
| - rtc::Buffer* packet,
|
| - const rtc::PacketTime& packet_time) {
|
| - const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
|
| - packet_time.not_before);
|
| - const webrtc::PacketReceiver::DeliveryStatus delivery_result =
|
| - call_->Receiver()->DeliverPacket(
|
| - webrtc::MediaType::VIDEO,
|
| - reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
|
| - webrtc_packet_time);
|
| - switch (delivery_result) {
|
| - case webrtc::PacketReceiver::DELIVERY_OK:
|
| - return;
|
| - case webrtc::PacketReceiver::DELIVERY_PACKET_ERROR:
|
| - return;
|
| - case webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC:
|
| - break;
|
| - }
|
| -
|
| - uint32_t ssrc = 0;
|
| - if (!GetRtpSsrc(packet->data(), packet->size(), &ssrc)) {
|
| - return;
|
| - }
|
| -
|
| - int payload_type = 0;
|
| - if (!GetRtpPayloadType(packet->data(), packet->size(), &payload_type)) {
|
| - return;
|
| - }
|
| -
|
| - // See if this payload_type is registered as one that usually gets its own
|
| - // SSRC (RTX) or at least is safe to drop either way (ULPFEC). If it is, and
|
| - // it wasn't handled above by DeliverPacket, that means we don't know what
|
| - // stream it associates with, and we shouldn't ever create an implicit channel
|
| - // for these.
|
| - for (auto& codec : recv_codecs_) {
|
| - if (payload_type == codec.rtx_payload_type ||
|
| - payload_type == codec.fec.red_rtx_payload_type ||
|
| - payload_type == codec.fec.ulpfec_payload_type) {
|
| - return;
|
| - }
|
| - }
|
| -
|
| - switch (unsignalled_ssrc_handler_->OnUnsignalledSsrc(this, ssrc)) {
|
| - case UnsignalledSsrcHandler::kDropPacket:
|
| - return;
|
| - case UnsignalledSsrcHandler::kDeliverPacket:
|
| - break;
|
| - }
|
| -
|
| - if (call_->Receiver()->DeliverPacket(
|
| - webrtc::MediaType::VIDEO,
|
| - reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
|
| - webrtc_packet_time) != webrtc::PacketReceiver::DELIVERY_OK) {
|
| - LOG(LS_WARNING) << "Failed to deliver RTP packet on re-delivery.";
|
| - return;
|
| - }
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::OnRtcpReceived(
|
| - rtc::Buffer* packet,
|
| - const rtc::PacketTime& packet_time) {
|
| - const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
|
| - packet_time.not_before);
|
| - // TODO(pbos): Check webrtc::PacketReceiver::DELIVERY_OK once we deliver
|
| - // for both audio and video on the same path. Since BundleFilter doesn't
|
| - // filter RTCP anymore incoming RTCP packets could've been going to audio (so
|
| - // logging failures spam the log).
|
| - call_->Receiver()->DeliverPacket(
|
| - webrtc::MediaType::VIDEO,
|
| - reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
|
| - webrtc_packet_time);
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::OnReadyToSend(bool ready) {
|
| - LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready.");
|
| - call_->SignalNetworkState(ready ? webrtc::kNetworkUp : webrtc::kNetworkDown);
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::MuteStream(uint32_t ssrc, bool mute) {
|
| - LOG(LS_VERBOSE) << "MuteStream: " << ssrc << " -> "
|
| - << (mute ? "mute" : "unmute");
|
| - RTC_DCHECK(ssrc != 0);
|
| - rtc::CritScope stream_lock(&stream_crit_);
|
| - if (send_streams_.find(ssrc) == send_streams_.end()) {
|
| - LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc;
|
| - return false;
|
| - }
|
| -
|
| - send_streams_[ssrc]->MuteStream(mute);
|
| - return true;
|
| -}
|
| -
|
| -// TODO(pbos): Remove SetOptions in favor of SetSendParameters.
|
| -void WebRtcVideoChannel2::SetOptions(const VideoOptions& options) {
|
| - VideoSendParameters new_params = send_params_;
|
| - new_params.options.SetAll(options);
|
| - SetSendParameters(send_params_);
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::SetInterface(NetworkInterface* iface) {
|
| - MediaChannel::SetInterface(iface);
|
| - // Set the RTP recv/send buffer to a bigger size
|
| - MediaChannel::SetOption(NetworkInterface::ST_RTP,
|
| - rtc::Socket::OPT_RCVBUF,
|
| - kVideoRtpBufferSize);
|
| -
|
| - // Speculative change to increase the outbound socket buffer size.
|
| - // In b/15152257, we are seeing a significant number of packets discarded
|
| - // due to lack of socket buffer space, although it's not yet clear what the
|
| - // ideal value should be.
|
| - MediaChannel::SetOption(NetworkInterface::ST_RTP,
|
| - rtc::Socket::OPT_SNDBUF,
|
| - kVideoRtpBufferSize);
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::OnLoadUpdate(Load load) {
|
| - // OnLoadUpdate can not take any locks that are held while creating streams
|
| - // etc. Doing so establishes lock-order inversions between the webrtc process
|
| - // thread on stream creation and locks such as stream_crit_ while calling out.
|
| - rtc::CritScope stream_lock(&capturer_crit_);
|
| - if (!signal_cpu_adaptation_)
|
| - return;
|
| - // Do not adapt resolution for screen content as this will likely result in
|
| - // blurry and unreadable text.
|
| - for (auto& kv : capturers_) {
|
| - if (kv.second != nullptr
|
| - && !kv.second->IsScreencast()
|
| - && kv.second->video_adapter() != nullptr) {
|
| - kv.second->video_adapter()->OnCpuResolutionRequest(
|
| - load == kOveruse ? CoordinatedVideoAdapter::DOWNGRADE
|
| - : CoordinatedVideoAdapter::UPGRADE);
|
| - }
|
| - }
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::SendRtp(const uint8_t* data,
|
| - size_t len,
|
| - const webrtc::PacketOptions& options) {
|
| - rtc::Buffer packet(data, len, kMaxRtpPacketLen);
|
| - rtc::PacketOptions rtc_options;
|
| - rtc_options.packet_id = options.packet_id;
|
| - return MediaChannel::SendPacket(&packet, rtc_options);
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::SendRtcp(const uint8_t* data, size_t len) {
|
| - rtc::Buffer packet(data, len, kMaxRtpPacketLen);
|
| - return MediaChannel::SendRtcp(&packet, rtc::PacketOptions());
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::StartAllSendStreams() {
|
| - rtc::CritScope stream_lock(&stream_crit_);
|
| - for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator it =
|
| - send_streams_.begin();
|
| - it != send_streams_.end(); ++it) {
|
| - it->second->Start();
|
| - }
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::StopAllSendStreams() {
|
| - rtc::CritScope stream_lock(&stream_crit_);
|
| - for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator it =
|
| - send_streams_.begin();
|
| - it != send_streams_.end(); ++it) {
|
| - it->second->Stop();
|
| - }
|
| -}
|
| -
|
| -WebRtcVideoChannel2::WebRtcVideoSendStream::VideoSendStreamParameters::
|
| - VideoSendStreamParameters(
|
| - const webrtc::VideoSendStream::Config& config,
|
| - const VideoOptions& options,
|
| - int max_bitrate_bps,
|
| - const rtc::Optional<VideoCodecSettings>& codec_settings)
|
| - : config(config),
|
| - options(options),
|
| - max_bitrate_bps(max_bitrate_bps),
|
| - codec_settings(codec_settings) {}
|
| -
|
| -WebRtcVideoChannel2::WebRtcVideoSendStream::AllocatedEncoder::AllocatedEncoder(
|
| - webrtc::VideoEncoder* encoder,
|
| - webrtc::VideoCodecType type,
|
| - bool external)
|
| - : encoder(encoder),
|
| - external_encoder(nullptr),
|
| - type(type),
|
| - external(external) {
|
| - if (external) {
|
| - external_encoder = encoder;
|
| - this->encoder =
|
| - new webrtc::VideoEncoderSoftwareFallbackWrapper(type, encoder);
|
| - }
|
| -}
|
| -
|
| -WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream(
|
| - webrtc::Call* call,
|
| - const StreamParams& sp,
|
| - const webrtc::VideoSendStream::Config& config,
|
| - WebRtcVideoEncoderFactory* external_encoder_factory,
|
| - const VideoOptions& options,
|
| - int max_bitrate_bps,
|
| - const rtc::Optional<VideoCodecSettings>& codec_settings,
|
| - const std::vector<webrtc::RtpExtension>& rtp_extensions,
|
| - // TODO(deadbeef): Don't duplicate information between send_params,
|
| - // rtp_extensions, options, etc.
|
| - const VideoSendParameters& send_params)
|
| - : ssrcs_(sp.ssrcs),
|
| - ssrc_groups_(sp.ssrc_groups),
|
| - call_(call),
|
| - external_encoder_factory_(external_encoder_factory),
|
| - stream_(NULL),
|
| - parameters_(config, options, max_bitrate_bps, codec_settings),
|
| - pending_encoder_reconfiguration_(false),
|
| - allocated_encoder_(NULL, webrtc::kVideoCodecUnknown, false),
|
| - capturer_(NULL),
|
| - sending_(false),
|
| - muted_(false),
|
| - old_adapt_changes_(0),
|
| - first_frame_timestamp_ms_(0),
|
| - last_frame_timestamp_ms_(0) {
|
| - parameters_.config.rtp.max_packet_size = kVideoMtu;
|
| -
|
| - sp.GetPrimarySsrcs(¶meters_.config.rtp.ssrcs);
|
| - sp.GetFidSsrcs(parameters_.config.rtp.ssrcs,
|
| - ¶meters_.config.rtp.rtx.ssrcs);
|
| - parameters_.config.rtp.c_name = sp.cname;
|
| - parameters_.config.rtp.extensions = rtp_extensions;
|
| - parameters_.config.rtp.rtcp_mode = send_params.rtcp.reduced_size
|
| - ? webrtc::RtcpMode::kReducedSize
|
| - : webrtc::RtcpMode::kCompound;
|
| -
|
| - if (codec_settings) {
|
| - SetCodecAndOptions(*codec_settings, parameters_.options);
|
| - }
|
| -}
|
| -
|
| -WebRtcVideoChannel2::WebRtcVideoSendStream::~WebRtcVideoSendStream() {
|
| - DisconnectCapturer();
|
| - if (stream_ != NULL) {
|
| - call_->DestroyVideoSendStream(stream_);
|
| - }
|
| - DestroyVideoEncoder(&allocated_encoder_);
|
| -}
|
| -
|
| -static void CreateBlackFrame(webrtc::VideoFrame* video_frame,
|
| - int width,
|
| - int height) {
|
| - video_frame->CreateEmptyFrame(width, height, width, (width + 1) / 2,
|
| - (width + 1) / 2);
|
| - memset(video_frame->buffer(webrtc::kYPlane), 16,
|
| - video_frame->allocated_size(webrtc::kYPlane));
|
| - memset(video_frame->buffer(webrtc::kUPlane), 128,
|
| - video_frame->allocated_size(webrtc::kUPlane));
|
| - memset(video_frame->buffer(webrtc::kVPlane), 128,
|
| - video_frame->allocated_size(webrtc::kVPlane));
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame(
|
| - VideoCapturer* capturer,
|
| - const VideoFrame* frame) {
|
| - TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::InputFrame");
|
| - webrtc::VideoFrame video_frame(frame->GetVideoFrameBuffer(), 0, 0,
|
| - frame->GetVideoRotation());
|
| - rtc::CritScope cs(&lock_);
|
| - if (stream_ == NULL) {
|
| - // Frame input before send codecs are configured, dropping frame.
|
| - return;
|
| - }
|
| -
|
| - // Not sending, abort early to prevent expensive reconfigurations while
|
| - // setting up codecs etc.
|
| - if (!sending_)
|
| - return;
|
| -
|
| - if (format_.width == 0) { // Dropping frames.
|
| - RTC_DCHECK(format_.height == 0);
|
| - LOG(LS_VERBOSE) << "VideoFormat 0x0 set, Dropping frame.";
|
| - return;
|
| - }
|
| - if (muted_) {
|
| - // Create a black frame to transmit instead.
|
| - CreateBlackFrame(&video_frame,
|
| - static_cast<int>(frame->GetWidth()),
|
| - static_cast<int>(frame->GetHeight()));
|
| - }
|
| -
|
| - int64_t frame_delta_ms = frame->GetTimeStamp() / rtc::kNumNanosecsPerMillisec;
|
| - // frame->GetTimeStamp() is essentially a delta, align to webrtc time
|
| - if (first_frame_timestamp_ms_ == 0) {
|
| - first_frame_timestamp_ms_ = rtc::Time() - frame_delta_ms;
|
| - }
|
| -
|
| - last_frame_timestamp_ms_ = first_frame_timestamp_ms_ + frame_delta_ms;
|
| - video_frame.set_render_time_ms(last_frame_timestamp_ms_);
|
| - // Reconfigure codec if necessary.
|
| - SetDimensions(
|
| - video_frame.width(), video_frame.height(), capturer->IsScreencast());
|
| -
|
| - stream_->Input()->IncomingCapturedFrame(video_frame);
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetCapturer(
|
| - VideoCapturer* capturer) {
|
| - TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::SetCapturer");
|
| - if (!DisconnectCapturer() && capturer == NULL) {
|
| - return false;
|
| - }
|
| -
|
| - {
|
| - rtc::CritScope cs(&lock_);
|
| -
|
| - // Reset timestamps to realign new incoming frames to a webrtc timestamp. A
|
| - // new capturer may have a different timestamp delta than the previous one.
|
| - first_frame_timestamp_ms_ = 0;
|
| -
|
| - if (capturer == NULL) {
|
| - if (stream_ != NULL) {
|
| - LOG(LS_VERBOSE) << "Disabling capturer, sending black frame.";
|
| - webrtc::VideoFrame black_frame;
|
| -
|
| - CreateBlackFrame(&black_frame, last_dimensions_.width,
|
| - last_dimensions_.height);
|
| -
|
| - // Force this black frame not to be dropped due to timestamp order
|
| - // check. As IncomingCapturedFrame will drop the frame if this frame's
|
| - // timestamp is less than or equal to last frame's timestamp, it is
|
| - // necessary to give this black frame a larger timestamp than the
|
| - // previous one.
|
| - last_frame_timestamp_ms_ +=
|
| - format_.interval / rtc::kNumNanosecsPerMillisec;
|
| - black_frame.set_render_time_ms(last_frame_timestamp_ms_);
|
| - stream_->Input()->IncomingCapturedFrame(black_frame);
|
| - }
|
| -
|
| - capturer_ = NULL;
|
| - return true;
|
| - }
|
| -
|
| - capturer_ = capturer;
|
| - }
|
| - // Lock cannot be held while connecting the capturer to prevent lock-order
|
| - // violations.
|
| - capturer->SignalVideoFrame.connect(this, &WebRtcVideoSendStream::InputFrame);
|
| - return true;
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::WebRtcVideoSendStream::MuteStream(bool mute) {
|
| - rtc::CritScope cs(&lock_);
|
| - muted_ = mute;
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::WebRtcVideoSendStream::DisconnectCapturer() {
|
| - cricket::VideoCapturer* capturer;
|
| - {
|
| - rtc::CritScope cs(&lock_);
|
| - if (capturer_ == NULL)
|
| - return false;
|
| -
|
| - if (capturer_->video_adapter() != nullptr)
|
| - old_adapt_changes_ += capturer_->video_adapter()->adaptation_changes();
|
| -
|
| - capturer = capturer_;
|
| - capturer_ = NULL;
|
| - }
|
| - capturer->SignalVideoFrame.disconnect(this);
|
| - return true;
|
| -}
|
| -
|
| -const std::vector<uint32_t>&
|
| -WebRtcVideoChannel2::WebRtcVideoSendStream::GetSsrcs() const {
|
| - return ssrcs_;
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::WebRtcVideoSendStream::SetOptions(
|
| - const VideoOptions& options) {
|
| - rtc::CritScope cs(&lock_);
|
| - if (parameters_.codec_settings) {
|
| - LOG(LS_INFO) << "SetCodecAndOptions because of SetOptions; options="
|
| - << options.ToString();
|
| - SetCodecAndOptions(*parameters_.codec_settings, options);
|
| - } else {
|
| - parameters_.options = options;
|
| - }
|
| -}
|
| -
|
| -webrtc::VideoCodecType CodecTypeFromName(const std::string& name) {
|
| - if (CodecNamesEq(name, kVp8CodecName)) {
|
| - return webrtc::kVideoCodecVP8;
|
| - } else if (CodecNamesEq(name, kVp9CodecName)) {
|
| - return webrtc::kVideoCodecVP9;
|
| - } else if (CodecNamesEq(name, kH264CodecName)) {
|
| - return webrtc::kVideoCodecH264;
|
| - }
|
| - return webrtc::kVideoCodecUnknown;
|
| -}
|
| -
|
| -WebRtcVideoChannel2::WebRtcVideoSendStream::AllocatedEncoder
|
| -WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoder(
|
| - const VideoCodec& codec) {
|
| - webrtc::VideoCodecType type = CodecTypeFromName(codec.name);
|
| -
|
| - // Do not re-create encoders of the same type.
|
| - if (type == allocated_encoder_.type && allocated_encoder_.encoder != NULL) {
|
| - return allocated_encoder_;
|
| - }
|
| -
|
| - if (external_encoder_factory_ != NULL) {
|
| - webrtc::VideoEncoder* encoder =
|
| - external_encoder_factory_->CreateVideoEncoder(type);
|
| - if (encoder != NULL) {
|
| - return AllocatedEncoder(encoder, type, true);
|
| - }
|
| - }
|
| -
|
| - if (type == webrtc::kVideoCodecVP8) {
|
| - return AllocatedEncoder(
|
| - webrtc::VideoEncoder::Create(webrtc::VideoEncoder::kVp8), type, false);
|
| - } else if (type == webrtc::kVideoCodecVP9) {
|
| - return AllocatedEncoder(
|
| - webrtc::VideoEncoder::Create(webrtc::VideoEncoder::kVp9), type, false);
|
| - } else if (type == webrtc::kVideoCodecH264) {
|
| - return AllocatedEncoder(
|
| - webrtc::VideoEncoder::Create(webrtc::VideoEncoder::kH264), type, false);
|
| - }
|
| -
|
| - // This shouldn't happen, we should not be trying to create something we don't
|
| - // support.
|
| - RTC_DCHECK(false);
|
| - return AllocatedEncoder(NULL, webrtc::kVideoCodecUnknown, false);
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::WebRtcVideoSendStream::DestroyVideoEncoder(
|
| - AllocatedEncoder* encoder) {
|
| - if (encoder->external) {
|
| - external_encoder_factory_->DestroyVideoEncoder(encoder->external_encoder);
|
| - }
|
| - delete encoder->encoder;
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::WebRtcVideoSendStream::SetCodecAndOptions(
|
| - const VideoCodecSettings& codec_settings,
|
| - const VideoOptions& options) {
|
| - parameters_.encoder_config =
|
| - CreateVideoEncoderConfig(last_dimensions_, codec_settings.codec);
|
| - RTC_DCHECK(!parameters_.encoder_config.streams.empty());
|
| -
|
| - format_ = VideoFormat(codec_settings.codec.width,
|
| - codec_settings.codec.height,
|
| - VideoFormat::FpsToInterval(30),
|
| - FOURCC_I420);
|
| -
|
| - AllocatedEncoder new_encoder = CreateVideoEncoder(codec_settings.codec);
|
| - parameters_.config.encoder_settings.encoder = new_encoder.encoder;
|
| - parameters_.config.encoder_settings.payload_name = codec_settings.codec.name;
|
| - parameters_.config.encoder_settings.payload_type = codec_settings.codec.id;
|
| - if (new_encoder.external) {
|
| - webrtc::VideoCodecType type = CodecTypeFromName(codec_settings.codec.name);
|
| - parameters_.config.encoder_settings.internal_source =
|
| - external_encoder_factory_->EncoderTypeHasInternalSource(type);
|
| - }
|
| - parameters_.config.rtp.fec = codec_settings.fec;
|
| -
|
| - // Set RTX payload type if RTX is enabled.
|
| - if (!parameters_.config.rtp.rtx.ssrcs.empty()) {
|
| - if (codec_settings.rtx_payload_type == -1) {
|
| - LOG(LS_WARNING) << "RTX SSRCs configured but there's no configured RTX "
|
| - "payload type. Ignoring.";
|
| - parameters_.config.rtp.rtx.ssrcs.clear();
|
| - } else {
|
| - parameters_.config.rtp.rtx.payload_type = codec_settings.rtx_payload_type;
|
| - }
|
| - }
|
| -
|
| - parameters_.config.rtp.nack.rtp_history_ms =
|
| - HasNack(codec_settings.codec) ? kNackHistoryMs : 0;
|
| -
|
| - RTC_CHECK(options.suspend_below_min_bitrate);
|
| - parameters_.config.suspend_below_min_bitrate =
|
| - *options.suspend_below_min_bitrate;
|
| -
|
| - parameters_.codec_settings =
|
| - rtc::Optional<WebRtcVideoChannel2::VideoCodecSettings>(codec_settings);
|
| - parameters_.options = options;
|
| -
|
| - LOG(LS_INFO)
|
| - << "RecreateWebRtcStream (send) because of SetCodecAndOptions; options="
|
| - << options.ToString();
|
| - RecreateWebRtcStream();
|
| - if (allocated_encoder_.encoder != new_encoder.encoder) {
|
| - DestroyVideoEncoder(&allocated_encoder_);
|
| - allocated_encoder_ = new_encoder;
|
| - }
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::WebRtcVideoSendStream::SetSendParameters(
|
| - const ChangedSendParameters& params) {
|
| - rtc::CritScope cs(&lock_);
|
| - // |recreate_stream| means construction-time parameters have changed and the
|
| - // sending stream needs to be reset with the new config.
|
| - bool recreate_stream = false;
|
| - if (params.rtcp_mode) {
|
| - parameters_.config.rtp.rtcp_mode = *params.rtcp_mode;
|
| - recreate_stream = true;
|
| - }
|
| - if (params.rtp_header_extensions) {
|
| - parameters_.config.rtp.extensions = *params.rtp_header_extensions;
|
| - if (capturer_) {
|
| - capturer_->SetApplyRotation(!ContainsHeaderExtension(
|
| - *params.rtp_header_extensions, kRtpVideoRotationHeaderExtension));
|
| - }
|
| - recreate_stream = true;
|
| - }
|
| - if (params.max_bandwidth_bps) {
|
| - // Max bitrate has changed, reconfigure encoder settings on the next frame
|
| - // or stream recreation.
|
| - parameters_.max_bitrate_bps = *params.max_bandwidth_bps;
|
| - pending_encoder_reconfiguration_ = true;
|
| - }
|
| - // Set codecs and options.
|
| - if (params.codec) {
|
| - SetCodecAndOptions(*params.codec,
|
| - params.options ? *params.options : parameters_.options);
|
| - return;
|
| - } else if (params.options) {
|
| - // Reconfigure if codecs are already set.
|
| - if (parameters_.codec_settings) {
|
| - SetCodecAndOptions(*parameters_.codec_settings, *params.options);
|
| - return;
|
| - } else {
|
| - parameters_.options = *params.options;
|
| - }
|
| - }
|
| - if (recreate_stream) {
|
| - LOG(LS_INFO) << "RecreateWebRtcStream (send) because of SetSendParameters";
|
| - RecreateWebRtcStream();
|
| - }
|
| -}
|
| -
|
| -webrtc::VideoEncoderConfig
|
| -WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig(
|
| - const Dimensions& dimensions,
|
| - const VideoCodec& codec) const {
|
| - webrtc::VideoEncoderConfig encoder_config;
|
| - if (dimensions.is_screencast) {
|
| - RTC_CHECK(parameters_.options.screencast_min_bitrate_kbps);
|
| - encoder_config.min_transmit_bitrate_bps =
|
| - *parameters_.options.screencast_min_bitrate_kbps * 1000;
|
| - encoder_config.content_type =
|
| - webrtc::VideoEncoderConfig::ContentType::kScreen;
|
| - } else {
|
| - encoder_config.min_transmit_bitrate_bps = 0;
|
| - encoder_config.content_type =
|
| - webrtc::VideoEncoderConfig::ContentType::kRealtimeVideo;
|
| - }
|
| -
|
| - // Restrict dimensions according to codec max.
|
| - int width = dimensions.width;
|
| - int height = dimensions.height;
|
| - if (!dimensions.is_screencast) {
|
| - if (codec.width < width)
|
| - width = codec.width;
|
| - if (codec.height < height)
|
| - height = codec.height;
|
| - }
|
| -
|
| - VideoCodec clamped_codec = codec;
|
| - clamped_codec.width = width;
|
| - clamped_codec.height = height;
|
| -
|
| - // By default, the stream count for the codec configuration should match the
|
| - // number of negotiated ssrcs. But if the codec is blacklisted for simulcast
|
| - // or a screencast, only configure a single stream.
|
| - size_t stream_count = parameters_.config.rtp.ssrcs.size();
|
| - if (IsCodecBlacklistedForSimulcast(codec.name) || dimensions.is_screencast) {
|
| - stream_count = 1;
|
| - }
|
| -
|
| - encoder_config.streams =
|
| - CreateVideoStreams(clamped_codec, parameters_.options,
|
| - parameters_.max_bitrate_bps, stream_count);
|
| -
|
| - // Conference mode screencast uses 2 temporal layers split at 100kbit.
|
| - if (parameters_.options.conference_mode.value_or(false) &&
|
| - dimensions.is_screencast && encoder_config.streams.size() == 1) {
|
| - ScreenshareLayerConfig config = ScreenshareLayerConfig::GetDefault();
|
| -
|
| - // For screenshare in conference mode, tl0 and tl1 bitrates are piggybacked
|
| - // on the VideoCodec struct as target and max bitrates, respectively.
|
| - // See eg. webrtc::VP8EncoderImpl::SetRates().
|
| - encoder_config.streams[0].target_bitrate_bps =
|
| - config.tl0_bitrate_kbps * 1000;
|
| - encoder_config.streams[0].max_bitrate_bps = config.tl1_bitrate_kbps * 1000;
|
| - encoder_config.streams[0].temporal_layer_thresholds_bps.clear();
|
| - encoder_config.streams[0].temporal_layer_thresholds_bps.push_back(
|
| - config.tl0_bitrate_kbps * 1000);
|
| - }
|
| - return encoder_config;
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::WebRtcVideoSendStream::SetDimensions(
|
| - int width,
|
| - int height,
|
| - bool is_screencast) {
|
| - if (last_dimensions_.width == width && last_dimensions_.height == height &&
|
| - last_dimensions_.is_screencast == is_screencast &&
|
| - !pending_encoder_reconfiguration_) {
|
| - // Configured using the same parameters, do not reconfigure.
|
| - return;
|
| - }
|
| - LOG(LS_INFO) << "SetDimensions: " << width << "x" << height
|
| - << (is_screencast ? " (screencast)" : " (not screencast)");
|
| -
|
| - last_dimensions_.width = width;
|
| - last_dimensions_.height = height;
|
| - last_dimensions_.is_screencast = is_screencast;
|
| -
|
| - RTC_DCHECK(!parameters_.encoder_config.streams.empty());
|
| -
|
| - RTC_CHECK(parameters_.codec_settings);
|
| - VideoCodecSettings codec_settings = *parameters_.codec_settings;
|
| -
|
| - webrtc::VideoEncoderConfig encoder_config =
|
| - CreateVideoEncoderConfig(last_dimensions_, codec_settings.codec);
|
| -
|
| - encoder_config.encoder_specific_settings = ConfigureVideoEncoderSettings(
|
| - codec_settings.codec, parameters_.options, is_screencast);
|
| -
|
| - bool stream_reconfigured = stream_->ReconfigureVideoEncoder(encoder_config);
|
| -
|
| - encoder_config.encoder_specific_settings = NULL;
|
| - pending_encoder_reconfiguration_ = false;
|
| -
|
| - if (!stream_reconfigured) {
|
| - LOG(LS_WARNING) << "Failed to reconfigure video encoder for dimensions: "
|
| - << width << "x" << height;
|
| - return;
|
| - }
|
| -
|
| - parameters_.encoder_config = encoder_config;
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::WebRtcVideoSendStream::Start() {
|
| - rtc::CritScope cs(&lock_);
|
| - RTC_DCHECK(stream_ != NULL);
|
| - stream_->Start();
|
| - sending_ = true;
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::WebRtcVideoSendStream::Stop() {
|
| - rtc::CritScope cs(&lock_);
|
| - if (stream_ != NULL) {
|
| - stream_->Stop();
|
| - }
|
| - sending_ = false;
|
| -}
|
| -
|
| -VideoSenderInfo
|
| -WebRtcVideoChannel2::WebRtcVideoSendStream::GetVideoSenderInfo() {
|
| - VideoSenderInfo info;
|
| - webrtc::VideoSendStream::Stats stats;
|
| - {
|
| - rtc::CritScope cs(&lock_);
|
| - for (uint32_t ssrc : parameters_.config.rtp.ssrcs)
|
| - info.add_ssrc(ssrc);
|
| -
|
| - if (parameters_.codec_settings)
|
| - info.codec_name = parameters_.codec_settings->codec.name;
|
| - for (size_t i = 0; i < parameters_.encoder_config.streams.size(); ++i) {
|
| - if (i == parameters_.encoder_config.streams.size() - 1) {
|
| - info.preferred_bitrate +=
|
| - parameters_.encoder_config.streams[i].max_bitrate_bps;
|
| - } else {
|
| - info.preferred_bitrate +=
|
| - parameters_.encoder_config.streams[i].target_bitrate_bps;
|
| - }
|
| - }
|
| -
|
| - if (stream_ == NULL)
|
| - return info;
|
| -
|
| - stats = stream_->GetStats();
|
| -
|
| - info.adapt_changes = old_adapt_changes_;
|
| - info.adapt_reason = CoordinatedVideoAdapter::ADAPTREASON_NONE;
|
| -
|
| - if (capturer_ != NULL) {
|
| - if (!capturer_->IsMuted()) {
|
| - VideoFormat last_captured_frame_format;
|
| - capturer_->GetStats(&info.adapt_frame_drops, &info.effects_frame_drops,
|
| - &info.capturer_frame_time,
|
| - &last_captured_frame_format);
|
| - info.input_frame_width = last_captured_frame_format.width;
|
| - info.input_frame_height = last_captured_frame_format.height;
|
| - }
|
| - if (capturer_->video_adapter() != nullptr) {
|
| - info.adapt_changes += capturer_->video_adapter()->adaptation_changes();
|
| - info.adapt_reason = capturer_->video_adapter()->adapt_reason();
|
| - }
|
| - }
|
| - }
|
| -
|
| - // Get bandwidth limitation info from stream_->GetStats().
|
| - // Input resolution (output from video_adapter) can be further scaled down or
|
| - // higher video layer(s) can be dropped due to bitrate constraints.
|
| - // Note, adapt_changes only include changes from the video_adapter.
|
| - if (stats.bw_limited_resolution)
|
| - info.adapt_reason |= CoordinatedVideoAdapter::ADAPTREASON_BANDWIDTH;
|
| -
|
| - info.encoder_implementation_name = stats.encoder_implementation_name;
|
| - info.ssrc_groups = ssrc_groups_;
|
| - info.framerate_input = stats.input_frame_rate;
|
| - info.framerate_sent = stats.encode_frame_rate;
|
| - info.avg_encode_ms = stats.avg_encode_time_ms;
|
| - info.encode_usage_percent = stats.encode_usage_percent;
|
| -
|
| - info.nominal_bitrate = stats.media_bitrate_bps;
|
| -
|
| - info.send_frame_width = 0;
|
| - info.send_frame_height = 0;
|
| - for (std::map<uint32_t, webrtc::VideoSendStream::StreamStats>::iterator it =
|
| - stats.substreams.begin();
|
| - it != stats.substreams.end(); ++it) {
|
| - // TODO(pbos): Wire up additional stats, such as padding bytes.
|
| - webrtc::VideoSendStream::StreamStats stream_stats = it->second;
|
| - info.bytes_sent += stream_stats.rtp_stats.transmitted.payload_bytes +
|
| - stream_stats.rtp_stats.transmitted.header_bytes +
|
| - stream_stats.rtp_stats.transmitted.padding_bytes;
|
| - info.packets_sent += stream_stats.rtp_stats.transmitted.packets;
|
| - info.packets_lost += stream_stats.rtcp_stats.cumulative_lost;
|
| - if (stream_stats.width > info.send_frame_width)
|
| - info.send_frame_width = stream_stats.width;
|
| - if (stream_stats.height > info.send_frame_height)
|
| - info.send_frame_height = stream_stats.height;
|
| - info.firs_rcvd += stream_stats.rtcp_packet_type_counts.fir_packets;
|
| - info.nacks_rcvd += stream_stats.rtcp_packet_type_counts.nack_packets;
|
| - info.plis_rcvd += stream_stats.rtcp_packet_type_counts.pli_packets;
|
| - }
|
| -
|
| - if (!stats.substreams.empty()) {
|
| - // TODO(pbos): Report fraction lost per SSRC.
|
| - webrtc::VideoSendStream::StreamStats first_stream_stats =
|
| - stats.substreams.begin()->second;
|
| - info.fraction_lost =
|
| - static_cast<float>(first_stream_stats.rtcp_stats.fraction_lost) /
|
| - (1 << 8);
|
| - }
|
| -
|
| - return info;
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::WebRtcVideoSendStream::FillBandwidthEstimationInfo(
|
| - BandwidthEstimationInfo* bwe_info) {
|
| - rtc::CritScope cs(&lock_);
|
| - if (stream_ == NULL) {
|
| - return;
|
| - }
|
| - webrtc::VideoSendStream::Stats stats = stream_->GetStats();
|
| - for (std::map<uint32_t, webrtc::VideoSendStream::StreamStats>::iterator it =
|
| - stats.substreams.begin();
|
| - it != stats.substreams.end(); ++it) {
|
| - bwe_info->transmit_bitrate += it->second.total_bitrate_bps;
|
| - bwe_info->retransmit_bitrate += it->second.retransmit_bitrate_bps;
|
| - }
|
| - bwe_info->target_enc_bitrate += stats.target_media_bitrate_bps;
|
| - bwe_info->actual_enc_bitrate += stats.media_bitrate_bps;
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::WebRtcVideoSendStream::RecreateWebRtcStream() {
|
| - if (stream_ != NULL) {
|
| - call_->DestroyVideoSendStream(stream_);
|
| - }
|
| -
|
| - RTC_CHECK(parameters_.codec_settings);
|
| - parameters_.encoder_config.encoder_specific_settings =
|
| - ConfigureVideoEncoderSettings(
|
| - parameters_.codec_settings->codec, parameters_.options,
|
| - parameters_.encoder_config.content_type ==
|
| - webrtc::VideoEncoderConfig::ContentType::kScreen);
|
| -
|
| - webrtc::VideoSendStream::Config config = parameters_.config;
|
| - if (!config.rtp.rtx.ssrcs.empty() && config.rtp.rtx.payload_type == -1) {
|
| - LOG(LS_WARNING) << "RTX SSRCs configured but there's no configured RTX "
|
| - "payload type the set codec. Ignoring RTX.";
|
| - config.rtp.rtx.ssrcs.clear();
|
| - }
|
| - stream_ = call_->CreateVideoSendStream(config, parameters_.encoder_config);
|
| -
|
| - parameters_.encoder_config.encoder_specific_settings = NULL;
|
| - pending_encoder_reconfiguration_ = false;
|
| -
|
| - if (sending_) {
|
| - stream_->Start();
|
| - }
|
| -}
|
| -
|
| -WebRtcVideoChannel2::WebRtcVideoReceiveStream::WebRtcVideoReceiveStream(
|
| - webrtc::Call* call,
|
| - const StreamParams& sp,
|
| - const webrtc::VideoReceiveStream::Config& config,
|
| - WebRtcVideoDecoderFactory* external_decoder_factory,
|
| - bool default_stream,
|
| - const std::vector<VideoCodecSettings>& recv_codecs,
|
| - bool disable_prerenderer_smoothing)
|
| - : call_(call),
|
| - ssrcs_(sp.ssrcs),
|
| - ssrc_groups_(sp.ssrc_groups),
|
| - stream_(NULL),
|
| - default_stream_(default_stream),
|
| - config_(config),
|
| - external_decoder_factory_(external_decoder_factory),
|
| - disable_prerenderer_smoothing_(disable_prerenderer_smoothing),
|
| - sink_(NULL),
|
| - last_width_(-1),
|
| - last_height_(-1),
|
| - first_frame_timestamp_(-1),
|
| - estimated_remote_start_ntp_time_ms_(0) {
|
| - config_.renderer = this;
|
| - std::vector<AllocatedDecoder> old_decoders;
|
| - ConfigureCodecs(recv_codecs, &old_decoders);
|
| - RecreateWebRtcStream();
|
| - RTC_DCHECK(old_decoders.empty());
|
| -}
|
| -
|
| -WebRtcVideoChannel2::WebRtcVideoReceiveStream::AllocatedDecoder::
|
| - AllocatedDecoder(webrtc::VideoDecoder* decoder,
|
| - webrtc::VideoCodecType type,
|
| - bool external)
|
| - : decoder(decoder),
|
| - external_decoder(nullptr),
|
| - type(type),
|
| - external(external) {
|
| - if (external) {
|
| - external_decoder = decoder;
|
| - this->decoder =
|
| - new webrtc::VideoDecoderSoftwareFallbackWrapper(type, external_decoder);
|
| - }
|
| -}
|
| -
|
| -WebRtcVideoChannel2::WebRtcVideoReceiveStream::~WebRtcVideoReceiveStream() {
|
| - call_->DestroyVideoReceiveStream(stream_);
|
| - ClearDecoders(&allocated_decoders_);
|
| -}
|
| -
|
| -const std::vector<uint32_t>&
|
| -WebRtcVideoChannel2::WebRtcVideoReceiveStream::GetSsrcs() const {
|
| - return ssrcs_;
|
| -}
|
| -
|
| -WebRtcVideoChannel2::WebRtcVideoReceiveStream::AllocatedDecoder
|
| -WebRtcVideoChannel2::WebRtcVideoReceiveStream::CreateOrReuseVideoDecoder(
|
| - std::vector<AllocatedDecoder>* old_decoders,
|
| - const VideoCodec& codec) {
|
| - webrtc::VideoCodecType type = CodecTypeFromName(codec.name);
|
| -
|
| - for (size_t i = 0; i < old_decoders->size(); ++i) {
|
| - if ((*old_decoders)[i].type == type) {
|
| - AllocatedDecoder decoder = (*old_decoders)[i];
|
| - (*old_decoders)[i] = old_decoders->back();
|
| - old_decoders->pop_back();
|
| - return decoder;
|
| - }
|
| - }
|
| -
|
| - if (external_decoder_factory_ != NULL) {
|
| - webrtc::VideoDecoder* decoder =
|
| - external_decoder_factory_->CreateVideoDecoder(type);
|
| - if (decoder != NULL) {
|
| - return AllocatedDecoder(decoder, type, true);
|
| - }
|
| - }
|
| -
|
| - if (type == webrtc::kVideoCodecVP8) {
|
| - return AllocatedDecoder(
|
| - webrtc::VideoDecoder::Create(webrtc::VideoDecoder::kVp8), type, false);
|
| - }
|
| -
|
| - if (type == webrtc::kVideoCodecVP9) {
|
| - return AllocatedDecoder(
|
| - webrtc::VideoDecoder::Create(webrtc::VideoDecoder::kVp9), type, false);
|
| - }
|
| -
|
| - if (type == webrtc::kVideoCodecH264) {
|
| - return AllocatedDecoder(
|
| - webrtc::VideoDecoder::Create(webrtc::VideoDecoder::kH264), type, false);
|
| - }
|
| -
|
| - return AllocatedDecoder(
|
| - webrtc::VideoDecoder::Create(webrtc::VideoDecoder::kUnsupportedCodec),
|
| - webrtc::kVideoCodecUnknown, false);
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::WebRtcVideoReceiveStream::ConfigureCodecs(
|
| - const std::vector<VideoCodecSettings>& recv_codecs,
|
| - std::vector<AllocatedDecoder>* old_decoders) {
|
| - *old_decoders = allocated_decoders_;
|
| - allocated_decoders_.clear();
|
| - config_.decoders.clear();
|
| - for (size_t i = 0; i < recv_codecs.size(); ++i) {
|
| - AllocatedDecoder allocated_decoder =
|
| - CreateOrReuseVideoDecoder(old_decoders, recv_codecs[i].codec);
|
| - allocated_decoders_.push_back(allocated_decoder);
|
| -
|
| - webrtc::VideoReceiveStream::Decoder decoder;
|
| - decoder.decoder = allocated_decoder.decoder;
|
| - decoder.payload_type = recv_codecs[i].codec.id;
|
| - decoder.payload_name = recv_codecs[i].codec.name;
|
| - config_.decoders.push_back(decoder);
|
| - }
|
| -
|
| - // TODO(pbos): Reconfigure RTX based on incoming recv_codecs.
|
| - config_.rtp.fec = recv_codecs.front().fec;
|
| - config_.rtp.nack.rtp_history_ms =
|
| - HasNack(recv_codecs.begin()->codec) ? kNackHistoryMs : 0;
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetLocalSsrc(
|
| - uint32_t local_ssrc) {
|
| - // TODO(pbos): Consider turning this sanity check into a RTC_DCHECK. You
|
| - // should not be able to create a sender with the same SSRC as a receiver, but
|
| - // right now this can't be done due to unittests depending on receiving what
|
| - // they are sending from the same MediaChannel.
|
| - if (local_ssrc == config_.rtp.remote_ssrc) {
|
| - LOG(LS_INFO) << "Ignoring call to SetLocalSsrc because parameters are "
|
| - "unchanged; local_ssrc=" << local_ssrc;
|
| - return;
|
| - }
|
| -
|
| - config_.rtp.local_ssrc = local_ssrc;
|
| - LOG(LS_INFO)
|
| - << "RecreateWebRtcStream (recv) because of SetLocalSsrc; local_ssrc="
|
| - << local_ssrc;
|
| - RecreateWebRtcStream();
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetFeedbackParameters(
|
| - bool nack_enabled,
|
| - bool remb_enabled,
|
| - bool transport_cc_enabled) {
|
| - int nack_history_ms = nack_enabled ? kNackHistoryMs : 0;
|
| - if (config_.rtp.nack.rtp_history_ms == nack_history_ms &&
|
| - config_.rtp.remb == remb_enabled &&
|
| - config_.rtp.transport_cc == transport_cc_enabled) {
|
| - LOG(LS_INFO)
|
| - << "Ignoring call to SetFeedbackParameters because parameters are "
|
| - "unchanged; nack="
|
| - << nack_enabled << ", remb=" << remb_enabled
|
| - << ", transport_cc=" << transport_cc_enabled;
|
| - return;
|
| - }
|
| - config_.rtp.remb = remb_enabled;
|
| - config_.rtp.nack.rtp_history_ms = nack_history_ms;
|
| - config_.rtp.transport_cc = transport_cc_enabled;
|
| - LOG(LS_INFO)
|
| - << "RecreateWebRtcStream (recv) because of SetFeedbackParameters; nack="
|
| - << nack_enabled << ", remb=" << remb_enabled
|
| - << ", transport_cc=" << transport_cc_enabled;
|
| - RecreateWebRtcStream();
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetRecvParameters(
|
| - const ChangedRecvParameters& params) {
|
| - bool needs_recreation = false;
|
| - std::vector<AllocatedDecoder> old_decoders;
|
| - if (params.codec_settings) {
|
| - ConfigureCodecs(*params.codec_settings, &old_decoders);
|
| - needs_recreation = true;
|
| - }
|
| - if (params.rtp_header_extensions) {
|
| - config_.rtp.extensions = *params.rtp_header_extensions;
|
| - needs_recreation = true;
|
| - }
|
| - if (params.rtcp_mode) {
|
| - config_.rtp.rtcp_mode = *params.rtcp_mode;
|
| - needs_recreation = true;
|
| - }
|
| - if (needs_recreation) {
|
| - LOG(LS_INFO) << "RecreateWebRtcStream (recv) because of SetRecvParameters";
|
| - RecreateWebRtcStream();
|
| - ClearDecoders(&old_decoders);
|
| - }
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::WebRtcVideoReceiveStream::RecreateWebRtcStream() {
|
| - if (stream_ != NULL) {
|
| - call_->DestroyVideoReceiveStream(stream_);
|
| - }
|
| - stream_ = call_->CreateVideoReceiveStream(config_);
|
| - stream_->Start();
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::WebRtcVideoReceiveStream::ClearDecoders(
|
| - std::vector<AllocatedDecoder>* allocated_decoders) {
|
| - for (size_t i = 0; i < allocated_decoders->size(); ++i) {
|
| - if ((*allocated_decoders)[i].external) {
|
| - external_decoder_factory_->DestroyVideoDecoder(
|
| - (*allocated_decoders)[i].external_decoder);
|
| - }
|
| - delete (*allocated_decoders)[i].decoder;
|
| - }
|
| - allocated_decoders->clear();
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::WebRtcVideoReceiveStream::RenderFrame(
|
| - const webrtc::VideoFrame& frame,
|
| - int time_to_render_ms) {
|
| - rtc::CritScope crit(&sink_lock_);
|
| -
|
| - if (first_frame_timestamp_ < 0)
|
| - first_frame_timestamp_ = frame.timestamp();
|
| - int64_t rtp_time_elapsed_since_first_frame =
|
| - (timestamp_wraparound_handler_.Unwrap(frame.timestamp()) -
|
| - first_frame_timestamp_);
|
| - int64_t elapsed_time_ms = rtp_time_elapsed_since_first_frame /
|
| - (cricket::kVideoCodecClockrate / 1000);
|
| - if (frame.ntp_time_ms() > 0)
|
| - estimated_remote_start_ntp_time_ms_ = frame.ntp_time_ms() - elapsed_time_ms;
|
| -
|
| - if (sink_ == NULL) {
|
| - LOG(LS_WARNING) << "VideoReceiveStream not connected to a VideoSink.";
|
| - return;
|
| - }
|
| -
|
| - last_width_ = frame.width();
|
| - last_height_ = frame.height();
|
| -
|
| - const WebRtcVideoFrame render_frame(
|
| - frame.video_frame_buffer(),
|
| - frame.render_time_ms() * rtc::kNumNanosecsPerMillisec, frame.rotation());
|
| - sink_->OnFrame(render_frame);
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::WebRtcVideoReceiveStream::IsTextureSupported() const {
|
| - return true;
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::WebRtcVideoReceiveStream::SmoothsRenderedFrames()
|
| - const {
|
| - return disable_prerenderer_smoothing_;
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::WebRtcVideoReceiveStream::IsDefaultStream() const {
|
| - return default_stream_;
|
| -}
|
| -
|
| -void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetSink(
|
| - rtc::VideoSinkInterface<cricket::VideoFrame>* sink) {
|
| - rtc::CritScope crit(&sink_lock_);
|
| - sink_ = sink;
|
| -}
|
| -
|
| -std::string
|
| -WebRtcVideoChannel2::WebRtcVideoReceiveStream::GetCodecNameFromPayloadType(
|
| - int payload_type) {
|
| - for (const webrtc::VideoReceiveStream::Decoder& decoder : config_.decoders) {
|
| - if (decoder.payload_type == payload_type) {
|
| - return decoder.payload_name;
|
| - }
|
| - }
|
| - return "";
|
| -}
|
| -
|
| -VideoReceiverInfo
|
| -WebRtcVideoChannel2::WebRtcVideoReceiveStream::GetVideoReceiverInfo() {
|
| - VideoReceiverInfo info;
|
| - info.ssrc_groups = ssrc_groups_;
|
| - info.add_ssrc(config_.rtp.remote_ssrc);
|
| - webrtc::VideoReceiveStream::Stats stats = stream_->GetStats();
|
| - info.decoder_implementation_name = stats.decoder_implementation_name;
|
| - info.bytes_rcvd = stats.rtp_stats.transmitted.payload_bytes +
|
| - stats.rtp_stats.transmitted.header_bytes +
|
| - stats.rtp_stats.transmitted.padding_bytes;
|
| - info.packets_rcvd = stats.rtp_stats.transmitted.packets;
|
| - info.packets_lost = stats.rtcp_stats.cumulative_lost;
|
| - info.fraction_lost =
|
| - static_cast<float>(stats.rtcp_stats.fraction_lost) / (1 << 8);
|
| -
|
| - info.framerate_rcvd = stats.network_frame_rate;
|
| - info.framerate_decoded = stats.decode_frame_rate;
|
| - info.framerate_output = stats.render_frame_rate;
|
| -
|
| - {
|
| - rtc::CritScope frame_cs(&sink_lock_);
|
| - info.frame_width = last_width_;
|
| - info.frame_height = last_height_;
|
| - info.capture_start_ntp_time_ms = estimated_remote_start_ntp_time_ms_;
|
| - }
|
| -
|
| - info.decode_ms = stats.decode_ms;
|
| - info.max_decode_ms = stats.max_decode_ms;
|
| - info.current_delay_ms = stats.current_delay_ms;
|
| - info.target_delay_ms = stats.target_delay_ms;
|
| - info.jitter_buffer_ms = stats.jitter_buffer_ms;
|
| - info.min_playout_delay_ms = stats.min_playout_delay_ms;
|
| - info.render_delay_ms = stats.render_delay_ms;
|
| -
|
| - info.codec_name = GetCodecNameFromPayloadType(stats.current_payload_type);
|
| -
|
| - info.firs_sent = stats.rtcp_packet_type_counts.fir_packets;
|
| - info.plis_sent = stats.rtcp_packet_type_counts.pli_packets;
|
| - info.nacks_sent = stats.rtcp_packet_type_counts.nack_packets;
|
| -
|
| - return info;
|
| -}
|
| -
|
| -WebRtcVideoChannel2::VideoCodecSettings::VideoCodecSettings()
|
| - : rtx_payload_type(-1) {}
|
| -
|
| -bool WebRtcVideoChannel2::VideoCodecSettings::operator==(
|
| - const WebRtcVideoChannel2::VideoCodecSettings& other) const {
|
| - return codec == other.codec &&
|
| - fec.ulpfec_payload_type == other.fec.ulpfec_payload_type &&
|
| - fec.red_payload_type == other.fec.red_payload_type &&
|
| - fec.red_rtx_payload_type == other.fec.red_rtx_payload_type &&
|
| - rtx_payload_type == other.rtx_payload_type;
|
| -}
|
| -
|
| -bool WebRtcVideoChannel2::VideoCodecSettings::operator!=(
|
| - const WebRtcVideoChannel2::VideoCodecSettings& other) const {
|
| - return !(*this == other);
|
| -}
|
| -
|
| -std::vector<WebRtcVideoChannel2::VideoCodecSettings>
|
| -WebRtcVideoChannel2::MapCodecs(const std::vector<VideoCodec>& codecs) {
|
| - RTC_DCHECK(!codecs.empty());
|
| -
|
| - std::vector<VideoCodecSettings> video_codecs;
|
| - std::map<int, bool> payload_used;
|
| - std::map<int, VideoCodec::CodecType> payload_codec_type;
|
| - // |rtx_mapping| maps video payload type to rtx payload type.
|
| - std::map<int, int> rtx_mapping;
|
| -
|
| - webrtc::FecConfig fec_settings;
|
| -
|
| - for (size_t i = 0; i < codecs.size(); ++i) {
|
| - const VideoCodec& in_codec = codecs[i];
|
| - int payload_type = in_codec.id;
|
| -
|
| - if (payload_used[payload_type]) {
|
| - LOG(LS_ERROR) << "Payload type already registered: "
|
| - << in_codec.ToString();
|
| - return std::vector<VideoCodecSettings>();
|
| - }
|
| - payload_used[payload_type] = true;
|
| - payload_codec_type[payload_type] = in_codec.GetCodecType();
|
| -
|
| - switch (in_codec.GetCodecType()) {
|
| - case VideoCodec::CODEC_RED: {
|
| - // RED payload type, should not have duplicates.
|
| - RTC_DCHECK(fec_settings.red_payload_type == -1);
|
| - fec_settings.red_payload_type = in_codec.id;
|
| - continue;
|
| - }
|
| -
|
| - case VideoCodec::CODEC_ULPFEC: {
|
| - // ULPFEC payload type, should not have duplicates.
|
| - RTC_DCHECK(fec_settings.ulpfec_payload_type == -1);
|
| - fec_settings.ulpfec_payload_type = in_codec.id;
|
| - continue;
|
| - }
|
| -
|
| - case VideoCodec::CODEC_RTX: {
|
| - int associated_payload_type;
|
| - if (!in_codec.GetParam(kCodecParamAssociatedPayloadType,
|
| - &associated_payload_type) ||
|
| - !IsValidRtpPayloadType(associated_payload_type)) {
|
| - LOG(LS_ERROR)
|
| - << "RTX codec with invalid or no associated payload type: "
|
| - << in_codec.ToString();
|
| - return std::vector<VideoCodecSettings>();
|
| - }
|
| - rtx_mapping[associated_payload_type] = in_codec.id;
|
| - continue;
|
| - }
|
| -
|
| - case VideoCodec::CODEC_VIDEO:
|
| - break;
|
| - }
|
| -
|
| - video_codecs.push_back(VideoCodecSettings());
|
| - video_codecs.back().codec = in_codec;
|
| - }
|
| -
|
| - // One of these codecs should have been a video codec. Only having FEC
|
| - // parameters into this code is a logic error.
|
| - RTC_DCHECK(!video_codecs.empty());
|
| -
|
| - for (std::map<int, int>::const_iterator it = rtx_mapping.begin();
|
| - it != rtx_mapping.end();
|
| - ++it) {
|
| - if (!payload_used[it->first]) {
|
| - LOG(LS_ERROR) << "RTX mapped to payload not in codec list.";
|
| - return std::vector<VideoCodecSettings>();
|
| - }
|
| - if (payload_codec_type[it->first] != VideoCodec::CODEC_VIDEO &&
|
| - payload_codec_type[it->first] != VideoCodec::CODEC_RED) {
|
| - LOG(LS_ERROR) << "RTX not mapped to regular video codec or RED codec.";
|
| - return std::vector<VideoCodecSettings>();
|
| - }
|
| -
|
| - if (it->first == fec_settings.red_payload_type) {
|
| - fec_settings.red_rtx_payload_type = it->second;
|
| - }
|
| - }
|
| -
|
| - for (size_t i = 0; i < video_codecs.size(); ++i) {
|
| - video_codecs[i].fec = fec_settings;
|
| - if (rtx_mapping[video_codecs[i].codec.id] != 0 &&
|
| - rtx_mapping[video_codecs[i].codec.id] !=
|
| - fec_settings.red_payload_type) {
|
| - video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
|
| - }
|
| - }
|
| -
|
| - return video_codecs;
|
| -}
|
| -
|
| -} // namespace cricket
|
| -
|
| -#endif // HAVE_WEBRTC_VIDEO
|
|
|
Chromium Code Reviews
Issue 1587193006:
Move talk/media to webrtc/media (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master