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// Copyright 2016 The Chromium Authors
// 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/functional/bind.h"
#include "base/functional/callback.h"
#include "base/functional/callback_helpers.h"
#include "base/logging.h"
#include "base/task/single_thread_task_runner.h"
#include "remoting/proto/audio.pb.h"
#include "remoting/protocol/audio_stub.h"
namespace remoting::protocol {
WebrtcAudioSinkAdapter::WebrtcAudioSinkAdapter(
webrtc::scoped_refptr<webrtc::MediaStreamInterface> stream,
base::WeakPtr<AudioStub> audio_stub)
: task_runner_(base::SingleThreadTaskRunner::GetCurrentDefault()),
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 remoting::protocol
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