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// Copyright (c) 2012 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/host/audio_capturer_linux.h"
#include <fcntl.h>
#include <sys/stat.h>
#include <sys/types.h>
#include <unistd.h>
#include "base/eintr_wrapper.h"
#include "base/file_path.h"
#include "base/lazy_instance.h"
#include "base/logging.h"
#include "base/stl_util.h"
#include "remoting/proto/audio.pb.h"
#include "remoting/host/chromoting_host_context.h"
namespace remoting {
namespace {
// PulseAudio's module-pipe-sink must be configured to use the following
// parameters for the sink we read from.
const AudioPacket_SamplingRate kSamplingRate = AudioPacket::SAMPLING_RATE_44100;
const int kChannels = 2;
const int kBytesPerSample = 2;
// Read data from the pipe every 40ms.
const int kCapturingPeriodMs = 40;
#if !defined(F_SETPIPE_SZ)
// F_SETPIPE_SZ is supported only starting linux 2.6.35, but we want to be able
// to compile this code on machines with older kernel.
#define F_SETPIPE_SZ 1031
#endif // defined(F_SETPIPE_SZ)
// Pipename used to capture audio stream from.
// TODO(sergeyu): Pass this to AudioCapturerLinux constructor once we have
// Linux-specific DesktopEnvironmentFactory
base::LazyInstance<FilePath>::Leaky
g_audio_pipe_name = LAZY_INSTANCE_INITIALIZER;
const int IsPacketOfSilence(const std::string& data) {
const int64* int_buf = reinterpret_cast<const int64*>(data.data());
for (size_t i = 0; i < data.size() / sizeof(int64); i++) {
if (int_buf[i] != 0)
return false;
}
for (size_t i = data.size() - data.size() % sizeof(int64);
i < data.size(); i++) {
if (data.data()[i] != 0)
return false;
}
return true;
}
} // namespace
AudioCapturerLinux::AudioCapturerLinux(const FilePath& pipe_name) {
pipe_fd_ = HANDLE_EINTR(open(
pipe_name.value().c_str(), O_RDONLY | O_NONBLOCK));
if (pipe_fd_ < 0) {
LOG(ERROR) << "Failed to open " << pipe_name.value();
return;
}
// Set buffer size for the pipe to the double of what's required for samples
// of each capturing period.
int pipe_buffer_size = 2 * kCapturingPeriodMs * kSamplingRate * kChannels *
kBytesPerSample / base::Time::kMillisecondsPerSecond;
int result = HANDLE_EINTR(fcntl(pipe_fd_, F_SETPIPE_SZ, pipe_buffer_size));
if (result < 0) {
PLOG(ERROR) << "fcntl";
}
WaitForPipeReadable();
}
AudioCapturerLinux::~AudioCapturerLinux() {
}
bool AudioCapturerLinux::Start(const PacketCapturedCallback& callback) {
if (pipe_fd_ < 0)
return false;
callback_ = callback;
return true;
}
void AudioCapturerLinux::Stop() {
callback_.Reset();
}
bool AudioCapturerLinux::IsStarted() {
return !callback_.is_null();
}
void AudioCapturerLinux::OnFileCanReadWithoutBlocking(int fd) {
DCHECK_EQ(fd, pipe_fd_);
StartTimer();
}
void AudioCapturerLinux::OnFileCanWriteWithoutBlocking(int fd) {
NOTREACHED();
}
void AudioCapturerLinux::StartTimer() {
started_time_ = base::TimeTicks::Now();
last_capture_samples_ = 0;
timer_.Start(FROM_HERE, base::TimeDelta::FromMilliseconds(kCapturingPeriodMs),
this, &AudioCapturerLinux::DoCapture);
}
void AudioCapturerLinux::DoCapture() {
DCHECK_GT(pipe_fd_, 0);
// Calculate how much we need read from the pipe. Pulseaudio doesn't control
// how much data it writes to the pipe, so we need to pace the stream, so
// that we read the exact number of the samples per second we need.
base::TimeDelta stream_position = base::TimeTicks::Now() - started_time_;
int64 stream_position_samples = stream_position.InMilliseconds() *
kSamplingRate / base::Time::kMillisecondsPerSecond;
int64 samples_to_capture =
stream_position_samples - last_capture_samples_;
last_capture_samples_ = stream_position_samples;
int64 read_size =
samples_to_capture * kChannels * kBytesPerSample;
std::string data = left_over_bytes_;
int pos = data.size();
left_over_bytes_.clear();
data.resize(read_size);
while (pos < read_size) {
int read_result = HANDLE_EINTR(
read(pipe_fd_, string_as_array(&data) + pos, read_size - pos));
if (read_result >= 0) {
pos += read_result;
} else {
if (errno != EWOULDBLOCK && errno != EAGAIN)
PLOG(ERROR) << "read";
break;
}
}
if (pos == 0) {
WaitForPipeReadable();
return;
}
// Save any incomplete samples we've read for later. Each packet should
// contain integer number of samples.
int incomplete_samples_bytes = pos % (kChannels * kBytesPerSample);
left_over_bytes_.assign(data, pos - incomplete_samples_bytes,
incomplete_samples_bytes);
data.resize(pos - incomplete_samples_bytes);
if (callback_.is_null())
return;
if (IsPacketOfSilence(data))
return;
scoped_ptr<AudioPacket> packet(new AudioPacket());
packet->add_data(data);
packet->set_encoding(AudioPacket::ENCODING_RAW);
packet->set_sampling_rate(kSamplingRate);
packet->set_bytes_per_sample(AudioPacket::BYTES_PER_SAMPLE_2);
packet->set_channels(AudioPacket::CHANNELS_STEREO);
callback_.Run(packet.Pass());
}
void AudioCapturerLinux::WaitForPipeReadable() {
timer_.Stop();
MessageLoopForIO::current()->WatchFileDescriptor(
pipe_fd_, false, MessageLoopForIO::WATCH_READ,
&file_descriptor_watcher_, this);
}
void AudioCapturerLinux::SetPipeName(const FilePath& pipe_name) {
g_audio_pipe_name.Get() = pipe_name;
}
bool AudioCapturer::IsSupported() {
return !g_audio_pipe_name.Get().empty();
}
scoped_ptr<AudioCapturer> AudioCapturer::Create() {
FilePath path = g_audio_pipe_name.Get();
if (path.empty())
return scoped_ptr<AudioCapturer>();
return scoped_ptr<AudioCapturer>(new AudioCapturerLinux(path));
}
} // namespace remoting
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