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- // Copyright 2016 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 "remoting/protocol/webrtc_audio_sink_adapter.h"
- #include "base/bind.h"
- #include "base/callback.h"
- #include "base/callback_helpers.h"
- #include "base/logging.h"
- #include "remoting/proto/audio.pb.h"
- #include "remoting/protocol/audio_stub.h"
- namespace remoting {
- namespace protocol {
- WebrtcAudioSinkAdapter::WebrtcAudioSinkAdapter(
- rtc::scoped_refptr<webrtc::MediaStreamInterface> stream,
- base::WeakPtr<AudioStub> audio_stub)
- : task_runner_(base::ThreadTaskRunnerHandle::Get()),
- audio_stub_(audio_stub),
- media_stream_(std::move(stream)) {
- webrtc::AudioTrackVector audio_tracks = media_stream_->GetAudioTracks();
- // Caller must verify that the media stream contains audio tracks.
- DCHECK(!audio_tracks.empty());
- if (audio_tracks.size() > 1U) {
- LOG(WARNING) << "Received media stream with multiple audio tracks.";
- }
- audio_track_ = audio_tracks[0];
- audio_track_->GetSource()->AddSink(this);
- }
- WebrtcAudioSinkAdapter::~WebrtcAudioSinkAdapter() {
- audio_track_->GetSource()->RemoveSink(this);
- }
- void WebrtcAudioSinkAdapter::OnData(const void* audio_data,
- int bits_per_sample,
- int sample_rate,
- size_t number_of_channels,
- size_t number_of_frames) {
- std::unique_ptr<AudioPacket> audio_packet(new AudioPacket());
- audio_packet->set_encoding(AudioPacket::ENCODING_RAW);
- switch (sample_rate) {
- case 44100:
- audio_packet->set_sampling_rate(AudioPacket::SAMPLING_RATE_44100);
- break;
- case 48000:
- audio_packet->set_sampling_rate(AudioPacket::SAMPLING_RATE_48000);
- break;
- default:
- LOG(WARNING) << "Unsupported sampling rate: " << sample_rate;
- return;
- }
- if (bits_per_sample != 16) {
- LOG(WARNING) << "Unsupported bits/sample: " << bits_per_sample;
- return;
- }
- audio_packet->set_bytes_per_sample(AudioPacket::BYTES_PER_SAMPLE_2);
- if (number_of_channels != 2) {
- LOG(WARNING) << "Unsupported number of channels: " << number_of_channels;
- return;
- }
- audio_packet->set_channels(AudioPacket::CHANNELS_STEREO);
- size_t data_size =
- number_of_frames * number_of_channels * (bits_per_sample / 8);
- audio_packet->add_data(reinterpret_cast<const char*>(audio_data), data_size);
- task_runner_->PostTask(
- FROM_HERE, base::BindOnce(&AudioStub::ProcessAudioPacket, audio_stub_,
- std::move(audio_packet), base::OnceClosure()));
- }
- } // namespace protocol
- } // namespace remoting
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