| // Copyright 2013 The Chromium Authors. All rights reserved. |
| // Use of this source code is governed by a BSD-style license that can be |
| // found in the LICENSE file. |
| |
| #include "chrome/renderer/media/cast_rtp_stream.h" |
| |
| #include <algorithm> |
| |
| #include "base/bind.h" |
| #include "base/callback_helpers.h" |
| #include "base/command_line.h" |
| #include "base/logging.h" |
| #include "base/memory/weak_ptr.h" |
| #include "base/strings/stringprintf.h" |
| #include "base/sys_info.h" |
| #include "base/trace_event/trace_event.h" |
| #include "chrome/common/chrome_switches.h" |
| #include "chrome/renderer/media/cast_session.h" |
| #include "chrome/renderer/media/cast_udp_transport.h" |
| #include "content/public/renderer/media_stream_audio_sink.h" |
| #include "content/public/renderer/media_stream_video_sink.h" |
| #include "content/public/renderer/render_thread.h" |
| #include "content/public/renderer/video_encode_accelerator.h" |
| #include "media/audio/audio_parameters.h" |
| #include "media/base/audio_bus.h" |
| #include "media/base/audio_converter.h" |
| #include "media/base/audio_fifo.h" |
| #include "media/base/bind_to_current_loop.h" |
| #include "media/base/limits.h" |
| #include "media/base/video_frame.h" |
| #include "media/cast/cast_config.h" |
| #include "media/cast/cast_defines.h" |
| #include "media/cast/cast_sender.h" |
| #include "media/cast/net/cast_transport_config.h" |
| #include "third_party/WebKit/public/platform/WebMediaStreamSource.h" |
| #include "ui/gfx/geometry/size.h" |
| |
| using media::cast::AudioSenderConfig; |
| using media::cast::VideoSenderConfig; |
| |
| namespace { |
| |
| const char kCodecNameOpus[] = "OPUS"; |
| const char kCodecNameVp8[] = "VP8"; |
| const char kCodecNameH264[] = "H264"; |
| |
| // To convert from kilobits per second to bits to per second. |
| const int kBitrateMultiplier = 1000; |
| |
| CastRtpPayloadParams DefaultOpusPayload() { |
| CastRtpPayloadParams payload; |
| payload.payload_type = media::cast::kDefaultRtpAudioPayloadType; |
| payload.max_latency_ms = media::cast::kDefaultRtpMaxDelayMs; |
| payload.ssrc = 1; |
| payload.feedback_ssrc = 2; |
| payload.clock_rate = media::cast::kDefaultAudioSamplingRate; |
| // The value is 0 which means VBR. |
| payload.min_bitrate = payload.max_bitrate = |
| media::cast::kDefaultAudioEncoderBitrate; |
| payload.channels = 2; |
| payload.max_frame_rate = 100; // 10 ms audio frames |
| payload.codec_name = kCodecNameOpus; |
| return payload; |
| } |
| |
| CastRtpPayloadParams DefaultVp8Payload() { |
| CastRtpPayloadParams payload; |
| payload.payload_type = media::cast::kDefaultRtpVideoPayloadType; |
| payload.max_latency_ms = media::cast::kDefaultRtpMaxDelayMs; |
| payload.ssrc = 11; |
| payload.feedback_ssrc = 12; |
| payload.clock_rate = media::cast::kVideoFrequency; |
| payload.max_bitrate = media::cast::kDefaultMaxVideoBitRate; |
| payload.min_bitrate = media::cast::kDefaultMinVideoBitRate; |
| payload.channels = 1; |
| payload.max_frame_rate = media::cast::kDefaultMaxFrameRate; |
| payload.codec_name = kCodecNameVp8; |
| return payload; |
| } |
| |
| CastRtpPayloadParams DefaultH264Payload() { |
| CastRtpPayloadParams payload; |
| payload.payload_type = 96; |
| payload.max_latency_ms = media::cast::kDefaultRtpMaxDelayMs; |
| payload.ssrc = 11; |
| payload.feedback_ssrc = 12; |
| payload.clock_rate = media::cast::kVideoFrequency; |
| payload.max_bitrate = media::cast::kDefaultMaxVideoBitRate; |
| payload.min_bitrate = media::cast::kDefaultMinVideoBitRate; |
| payload.channels = 1; |
| payload.max_frame_rate = media::cast::kDefaultMaxFrameRate; |
| payload.codec_name = kCodecNameH264; |
| return payload; |
| } |
| |
| bool IsHardwareVP8EncodingSupported() { |
| const base::CommandLine* cmd_line = base::CommandLine::ForCurrentProcess(); |
| if (cmd_line->HasSwitch(switches::kDisableCastStreamingHWEncoding)) { |
| DVLOG(1) << "Disabled hardware VP8 support for Cast Streaming."; |
| return false; |
| } |
| |
| // Query for hardware VP8 encoder support. |
| const std::vector<media::VideoEncodeAccelerator::SupportedProfile> |
| vea_profiles = content::GetSupportedVideoEncodeAcceleratorProfiles(); |
| for (const auto& vea_profile : vea_profiles) { |
| if (vea_profile.profile >= media::VP8PROFILE_MIN && |
| vea_profile.profile <= media::VP8PROFILE_MAX) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| bool IsHardwareH264EncodingSupported() { |
| const base::CommandLine* cmd_line = base::CommandLine::ForCurrentProcess(); |
| if (cmd_line->HasSwitch(switches::kDisableCastStreamingHWEncoding)) { |
| DVLOG(1) << "Disabled hardware h264 support for Cast Streaming."; |
| return false; |
| } |
| |
| // Query for hardware H.264 encoder support. |
| const std::vector<media::VideoEncodeAccelerator::SupportedProfile> |
| vea_profiles = content::GetSupportedVideoEncodeAcceleratorProfiles(); |
| for (const auto& vea_profile : vea_profiles) { |
| if (vea_profile.profile >= media::H264PROFILE_MIN && |
| vea_profile.profile <= media::H264PROFILE_MAX) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| int NumberOfEncodeThreads() { |
| // Do not saturate CPU utilization just for encoding. On a lower-end system |
| // with only 1 or 2 cores, use only one thread for encoding. On systems with |
| // more cores, allow half of the cores to be used for encoding. |
| return std::min(8, (base::SysInfo::NumberOfProcessors() + 1) / 2); |
| } |
| |
| std::vector<CastRtpParams> SupportedAudioParams() { |
| // TODO(hclam): Fill in more codecs here. |
| return std::vector<CastRtpParams>(1, CastRtpParams(DefaultOpusPayload())); |
| } |
| |
| std::vector<CastRtpParams> SupportedVideoParams() { |
| std::vector<CastRtpParams> supported_params; |
| |
| // Prefer VP8 over H.264 for hardware encoder. |
| if (IsHardwareVP8EncodingSupported()) |
| supported_params.push_back(CastRtpParams(DefaultVp8Payload())); |
| if (IsHardwareH264EncodingSupported()) |
| supported_params.push_back(CastRtpParams(DefaultH264Payload())); |
| |
| // Propose the default software VP8 encoder, if no hardware encoders are |
| // available. |
| if (supported_params.empty()) |
| supported_params.push_back(CastRtpParams(DefaultVp8Payload())); |
| |
| return supported_params; |
| } |
| |
| bool ToAudioSenderConfig(const CastRtpParams& params, |
| AudioSenderConfig* config) { |
| config->ssrc = params.payload.ssrc; |
| config->receiver_ssrc = params.payload.feedback_ssrc; |
| if (config->ssrc == config->receiver_ssrc) |
| return false; |
| config->min_playout_delay = base::TimeDelta::FromMilliseconds( |
| params.payload.min_latency_ms ? |
| params.payload.min_latency_ms : |
| params.payload.max_latency_ms); |
| config->max_playout_delay = |
| base::TimeDelta::FromMilliseconds(params.payload.max_latency_ms); |
| if (config->min_playout_delay <= base::TimeDelta()) |
| return false; |
| if (config->min_playout_delay > config->max_playout_delay) |
| return false; |
| config->rtp_payload_type = params.payload.payload_type; |
| config->use_external_encoder = false; |
| config->frequency = params.payload.clock_rate; |
| if (config->frequency < media::cast::kMinSampleRateForEncoding) |
| return false; |
| config->channels = params.payload.channels; |
| if (config->channels < 1) |
| return false; |
| config->bitrate = params.payload.max_bitrate * kBitrateMultiplier; |
| if (params.payload.codec_name == kCodecNameOpus) |
| config->codec = media::cast::CODEC_AUDIO_OPUS; |
| else |
| return false; |
| config->aes_key = params.payload.aes_key; |
| config->aes_iv_mask = params.payload.aes_iv_mask; |
| return true; |
| } |
| |
| bool ToVideoSenderConfig(const CastRtpParams& params, |
| VideoSenderConfig* config) { |
| config->ssrc = params.payload.ssrc; |
| config->receiver_ssrc = params.payload.feedback_ssrc; |
| if (config->ssrc == config->receiver_ssrc) |
| return false; |
| config->min_playout_delay = base::TimeDelta::FromMilliseconds( |
| params.payload.min_latency_ms ? |
| params.payload.min_latency_ms : |
| params.payload.max_latency_ms); |
| config->max_playout_delay = |
| base::TimeDelta::FromMilliseconds(params.payload.max_latency_ms); |
| if (config->min_playout_delay <= base::TimeDelta()) |
| return false; |
| if (config->min_playout_delay > config->max_playout_delay) |
| return false; |
| config->rtp_payload_type = params.payload.payload_type; |
| config->min_bitrate = config->start_bitrate = |
| params.payload.min_bitrate * kBitrateMultiplier; |
| config->max_bitrate = params.payload.max_bitrate * kBitrateMultiplier; |
| if (config->min_bitrate > config->max_bitrate) |
| return false; |
| config->start_bitrate = config->min_bitrate; |
| config->max_frame_rate = static_cast<int>( |
| std::max(1.0, params.payload.max_frame_rate) + 0.5); |
| if (config->max_frame_rate > media::limits::kMaxFramesPerSecond) |
| return false; |
| if (params.payload.codec_name == kCodecNameVp8) { |
| config->use_external_encoder = IsHardwareVP8EncodingSupported(); |
| config->codec = media::cast::CODEC_VIDEO_VP8; |
| } else if (params.payload.codec_name == kCodecNameH264) { |
| config->use_external_encoder = IsHardwareH264EncodingSupported(); |
| config->codec = media::cast::CODEC_VIDEO_H264; |
| } else { |
| return false; |
| } |
| if (!config->use_external_encoder) |
| config->number_of_encode_threads = NumberOfEncodeThreads(); |
| config->aes_key = params.payload.aes_key; |
| config->aes_iv_mask = params.payload.aes_iv_mask; |
| return true; |
| } |
| |
| } // namespace |
| |
| // This class receives MediaStreamTrack events and video frames from a |
| // MediaStreamTrack. |
| // |
| // Threading: Video frames are received on the IO thread and then |
| // forwarded to media::cast::VideoFrameInput through a static method. |
| // Member variables of this class are only accessed on the render thread. |
| class CastVideoSink : public base::SupportsWeakPtr<CastVideoSink>, |
| public content::MediaStreamVideoSink { |
| public: |
| // |track| provides data for this sink. |
| // |error_callback| is called if video formats don't match. |
| CastVideoSink(const blink::WebMediaStreamTrack& track, |
| const CastRtpStream::ErrorCallback& error_callback) |
| : track_(track), |
| sink_added_(false), |
| error_callback_(error_callback) {} |
| |
| ~CastVideoSink() override { |
| if (sink_added_) |
| RemoveFromVideoTrack(this, track_); |
| } |
| |
| // This static method is used to forward video frames to |frame_input|. |
| static void OnVideoFrame( |
| // These parameters are already bound when callback is created. |
| const CastRtpStream::ErrorCallback& error_callback, |
| const scoped_refptr<media::cast::VideoFrameInput> frame_input, |
| // These parameters are passed for each frame. |
| const scoped_refptr<media::VideoFrame>& frame, |
| base::TimeTicks estimated_capture_time) { |
| const base::TimeTicks timestamp = estimated_capture_time.is_null() |
| ? base::TimeTicks::Now() |
| : estimated_capture_time; |
| |
| // Used by chrome/browser/extension/api/cast_streaming/performance_test.cc |
| TRACE_EVENT_INSTANT2( |
| "cast_perf_test", "MediaStreamVideoSink::OnVideoFrame", |
| TRACE_EVENT_SCOPE_THREAD, |
| "timestamp", timestamp.ToInternalValue(), |
| "time_delta", frame->timestamp().ToInternalValue()); |
| frame_input->InsertRawVideoFrame(frame, timestamp); |
| } |
| |
| // Attach this sink to a video track represented by |track_|. |
| // Data received from the track will be submitted to |frame_input|. |
| void AddToTrack( |
| const scoped_refptr<media::cast::VideoFrameInput>& frame_input) { |
| DCHECK(!sink_added_); |
| sink_added_ = true; |
| AddToVideoTrack( |
| this, |
| base::Bind( |
| &CastVideoSink::OnVideoFrame, |
| error_callback_, |
| frame_input), |
| track_); |
| } |
| |
| private: |
| blink::WebMediaStreamTrack track_; |
| bool sink_added_; |
| CastRtpStream::ErrorCallback error_callback_; |
| |
| DISALLOW_COPY_AND_ASSIGN(CastVideoSink); |
| }; |
| |
| // Receives audio data from a MediaStreamTrack. Data is submitted to |
| // media::cast::FrameInput. |
| // |
| // Threading: Audio frames are received on the real-time audio thread. |
| // Note that RemoveFromAudioTrack() is synchronous and we have |
| // gurantee that there will be no more audio data after calling it. |
| class CastAudioSink : public base::SupportsWeakPtr<CastAudioSink>, |
| public content::MediaStreamAudioSink, |
| public media::AudioConverter::InputCallback { |
| public: |
| // |track| provides data for this sink. |
| CastAudioSink(const blink::WebMediaStreamTrack& track, |
| int output_channels, |
| int output_sample_rate) |
| : track_(track), |
| output_channels_(output_channels), |
| output_sample_rate_(output_sample_rate), |
| current_input_bus_(nullptr), |
| sample_frames_in_(0), |
| sample_frames_out_(0) {} |
| |
| ~CastAudioSink() override { |
| if (frame_input_.get()) |
| RemoveFromAudioTrack(this, track_); |
| } |
| |
| // Add this sink to the track. Data received from the track will be |
| // submitted to |frame_input|. |
| void AddToTrack( |
| const scoped_refptr<media::cast::AudioFrameInput>& frame_input) { |
| DCHECK(frame_input.get()); |
| DCHECK(!frame_input_.get()); |
| // This member is written here and then accessed on the IO thread |
| // We will not get data until AddToAudioTrack is called so it is |
| // safe to access this member now. |
| frame_input_ = frame_input; |
| AddToAudioTrack(this, track_); |
| } |
| |
| protected: |
| // Called on real-time audio thread. |
| void OnData(const media::AudioBus& input_bus, |
| base::TimeTicks estimated_capture_time) override { |
| DCHECK(input_params_.IsValid()); |
| DCHECK_EQ(input_bus.channels(), input_params_.channels()); |
| DCHECK_EQ(input_bus.frames(), input_params_.frames_per_buffer()); |
| DCHECK(!estimated_capture_time.is_null()); |
| DCHECK(converter_.get()); |
| |
| // Determine the duration of the audio signal enqueued within |converter_|. |
| const base::TimeDelta signal_duration_already_buffered = |
| (sample_frames_in_ * base::TimeDelta::FromSeconds(1) / |
| input_params_.sample_rate()) - |
| (sample_frames_out_ * base::TimeDelta::FromSeconds(1) / |
| output_sample_rate_); |
| DVLOG(2) << "Audio reference time adjustment: -(" |
| << signal_duration_already_buffered.InMicroseconds() << " us)"; |
| const base::TimeTicks capture_time_of_first_converted_sample = |
| estimated_capture_time - signal_duration_already_buffered; |
| |
| // Convert the entire input signal. AudioConverter is efficient in that no |
| // additional copying or conversion will occur if the input signal is in the |
| // same format as the output. Note that, while the number of sample frames |
| // provided as input is always the same, the chunk size (and the size of the |
| // |audio_bus| here) can be variable. This is not an issue since |
| // media::cast::AudioFrameInput can handle variable-sized AudioBuses. |
| scoped_ptr<media::AudioBus> audio_bus = |
| media::AudioBus::Create(output_channels_, converter_->ChunkSize()); |
| // AudioConverter will call ProvideInput() to fetch from |current_data_|. |
| current_input_bus_ = &input_bus; |
| converter_->Convert(audio_bus.get()); |
| DCHECK(!current_input_bus_); // ProvideInput() called exactly once? |
| |
| sample_frames_in_ += input_params_.frames_per_buffer(); |
| sample_frames_out_ += audio_bus->frames(); |
| |
| frame_input_->InsertAudio(audio_bus.Pass(), |
| capture_time_of_first_converted_sample); |
| } |
| |
| // Called on real-time audio thread. |
| void OnSetFormat(const media::AudioParameters& params) override { |
| if (input_params_.Equals(params)) |
| return; |
| input_params_ = params; |
| |
| DVLOG(1) << "Setting up audio resampling: {" |
| << input_params_.channels() << " channels, " |
| << input_params_.sample_rate() << " Hz} --> {" |
| << output_channels_ << " channels, " |
| << output_sample_rate_ << " Hz}"; |
| const media::AudioParameters output_params( |
| media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
| media::GuessChannelLayout(output_channels_), |
| output_sample_rate_, 32, |
| output_sample_rate_ * input_params_.frames_per_buffer() / |
| input_params_.sample_rate()); |
| converter_.reset( |
| new media::AudioConverter(input_params_, output_params, false)); |
| converter_->AddInput(this); |
| sample_frames_in_ = 0; |
| sample_frames_out_ = 0; |
| } |
| |
| // Called on real-time audio thread. |
| double ProvideInput(media::AudioBus* audio_bus, |
| base::TimeDelta buffer_delay) override { |
| DCHECK(current_input_bus_); |
| current_input_bus_->CopyTo(audio_bus); |
| current_input_bus_ = nullptr; |
| return 1.0; |
| } |
| |
| private: |
| const blink::WebMediaStreamTrack track_; |
| const int output_channels_; |
| const int output_sample_rate_; |
| |
| // This must be set before the real-time audio thread starts calling OnData(), |
| // and remain unchanged until after the thread will stop calling OnData(). |
| scoped_refptr<media::cast::AudioFrameInput> frame_input_; |
| |
| // These members are accessed on the real-time audio time only. |
| media::AudioParameters input_params_; |
| scoped_ptr<media::AudioConverter> converter_; |
| const media::AudioBus* current_input_bus_; |
| int64 sample_frames_in_; |
| int64 sample_frames_out_; |
| |
| DISALLOW_COPY_AND_ASSIGN(CastAudioSink); |
| }; |
| |
| CastRtpParams::CastRtpParams(const CastRtpPayloadParams& payload_params) |
| : payload(payload_params) {} |
| |
| CastCodecSpecificParams::CastCodecSpecificParams() {} |
| |
| CastCodecSpecificParams::~CastCodecSpecificParams() {} |
| |
| CastRtpPayloadParams::CastRtpPayloadParams() |
| : payload_type(0), |
| max_latency_ms(0), |
| min_latency_ms(0), |
| ssrc(0), |
| feedback_ssrc(0), |
| clock_rate(0), |
| max_bitrate(0), |
| min_bitrate(0), |
| channels(0), |
| max_frame_rate(0.0) { |
| } |
| |
| CastRtpPayloadParams::~CastRtpPayloadParams() {} |
| |
| CastRtpParams::CastRtpParams() {} |
| |
| CastRtpParams::~CastRtpParams() {} |
| |
| CastRtpStream::CastRtpStream(const blink::WebMediaStreamTrack& track, |
| const scoped_refptr<CastSession>& session) |
| : track_(track), cast_session_(session), weak_factory_(this) {} |
| |
| CastRtpStream::~CastRtpStream() { |
| Stop(); |
| } |
| |
| std::vector<CastRtpParams> CastRtpStream::GetSupportedParams() { |
| if (IsAudio()) |
| return SupportedAudioParams(); |
| else |
| return SupportedVideoParams(); |
| } |
| |
| CastRtpParams CastRtpStream::GetParams() { return params_; } |
| |
| void CastRtpStream::Start(const CastRtpParams& params, |
| const base::Closure& start_callback, |
| const base::Closure& stop_callback, |
| const ErrorCallback& error_callback) { |
| DCHECK(!start_callback.is_null()); |
| DCHECK(!stop_callback.is_null()); |
| DCHECK(!error_callback.is_null()); |
| |
| DVLOG(1) << "CastRtpStream::Start = " << (IsAudio() ? "audio" : "video"); |
| stop_callback_ = stop_callback; |
| error_callback_ = error_callback; |
| |
| if (IsAudio()) { |
| AudioSenderConfig config; |
| if (!ToAudioSenderConfig(params, &config)) { |
| DidEncounterError("Invalid parameters for audio."); |
| return; |
| } |
| |
| // In case of error we have to go through DidEncounterError() to stop |
| // the streaming after reporting the error. |
| audio_sink_.reset(new CastAudioSink( |
| track_, |
| params.payload.channels, |
| params.payload.clock_rate)); |
| cast_session_->StartAudio( |
| config, |
| base::Bind(&CastAudioSink::AddToTrack, audio_sink_->AsWeakPtr()), |
| base::Bind(&CastRtpStream::DidEncounterError, |
| weak_factory_.GetWeakPtr())); |
| start_callback.Run(); |
| } else { |
| VideoSenderConfig config; |
| if (!ToVideoSenderConfig(params, &config)) { |
| DidEncounterError("Invalid parameters for video."); |
| return; |
| } |
| // See the code for audio above for explanation of callbacks. |
| video_sink_.reset(new CastVideoSink( |
| track_, |
| media::BindToCurrentLoop(base::Bind(&CastRtpStream::DidEncounterError, |
| weak_factory_.GetWeakPtr())))); |
| cast_session_->StartVideo( |
| config, |
| base::Bind(&CastVideoSink::AddToTrack, video_sink_->AsWeakPtr()), |
| base::Bind(&CastRtpStream::DidEncounterError, |
| weak_factory_.GetWeakPtr())); |
| start_callback.Run(); |
| } |
| } |
| |
| void CastRtpStream::Stop() { |
| DVLOG(1) << "CastRtpStream::Stop = " << (IsAudio() ? "audio" : "video"); |
| if (stop_callback_.is_null()) |
| return; // Already stopped. |
| weak_factory_.InvalidateWeakPtrs(); |
| error_callback_.Reset(); |
| audio_sink_.reset(); |
| video_sink_.reset(); |
| base::ResetAndReturn(&stop_callback_).Run(); |
| } |
| |
| void CastRtpStream::ToggleLogging(bool enable) { |
| DVLOG(1) << "CastRtpStream::ToggleLogging(" << enable << ") = " |
| << (IsAudio() ? "audio" : "video"); |
| cast_session_->ToggleLogging(IsAudio(), enable); |
| } |
| |
| void CastRtpStream::GetRawEvents( |
| const base::Callback<void(scoped_ptr<base::BinaryValue>)>& callback, |
| const std::string& extra_data) { |
| DVLOG(1) << "CastRtpStream::GetRawEvents = " |
| << (IsAudio() ? "audio" : "video"); |
| cast_session_->GetEventLogsAndReset(IsAudio(), extra_data, callback); |
| } |
| |
| void CastRtpStream::GetStats( |
| const base::Callback<void(scoped_ptr<base::DictionaryValue>)>& callback) { |
| DVLOG(1) << "CastRtpStream::GetStats = " |
| << (IsAudio() ? "audio" : "video"); |
| cast_session_->GetStatsAndReset(IsAudio(), callback); |
| } |
| |
| bool CastRtpStream::IsAudio() const { |
| return track_.source().type() == blink::WebMediaStreamSource::TypeAudio; |
| } |
| |
| void CastRtpStream::DidEncounterError(const std::string& message) { |
| DCHECK(content::RenderThread::Get()); |
| DVLOG(1) << "CastRtpStream::DidEncounterError(" << message << ") = " |
| << (IsAudio() ? "audio" : "video"); |
| // Save the WeakPtr first because the error callback might delete this object. |
| base::WeakPtr<CastRtpStream> ptr = weak_factory_.GetWeakPtr(); |
| error_callback_.Run(message); |
| base::ThreadTaskRunnerHandle::Get()->PostTask( |
| FROM_HERE, |
| base::Bind(&CastRtpStream::Stop, ptr)); |
| } |