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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 "content/renderer/media/audio_renderer_impl.h"
#include <math.h>
#include <algorithm>
#include "base/bind.h"
#include "content/common/child_process.h"
#include "content/common/media/audio_messages.h"
#include "content/renderer/render_thread_impl.h"
#include "media/audio/audio_buffers_state.h"
#include "media/audio/audio_util.h"
#include "media/base/filter_host.h"
AudioRendererImpl::AudioRendererImpl(media::AudioRendererSink* sink)
: AudioRendererBase(),
bytes_per_second_(0),
stopped_(false),
sink_(sink),
is_initialized_(false) {
}
AudioRendererImpl::~AudioRendererImpl() {
}
base::TimeDelta AudioRendererImpl::ConvertToDuration(int bytes) {
if (bytes_per_second_) {
return base::TimeDelta::FromMicroseconds(
base::Time::kMicrosecondsPerSecond * bytes / bytes_per_second_);
}
return base::TimeDelta();
}
void AudioRendererImpl::UpdateEarliestEndTime(int bytes_filled,
base::TimeDelta request_delay,
base::Time time_now) {
if (bytes_filled != 0) {
base::TimeDelta predicted_play_time = ConvertToDuration(bytes_filled);
float playback_rate = GetPlaybackRate();
if (playback_rate != 1.0f) {
predicted_play_time = base::TimeDelta::FromMicroseconds(
static_cast<int64>(ceil(predicted_play_time.InMicroseconds() *
playback_rate)));
}
earliest_end_time_ =
std::max(earliest_end_time_,
time_now + request_delay + predicted_play_time);
}
}
bool AudioRendererImpl::OnInitialize(int bits_per_channel,
ChannelLayout channel_layout,
int sample_rate) {
// We use the AUDIO_PCM_LINEAR flag because AUDIO_PCM_LOW_LATENCY
// does not currently support all the sample-rates that we require.
// Please see: http://code.google.com/p/chromium/issues/detail?id=103627
// for more details.
audio_parameters_ = AudioParameters(AudioParameters::AUDIO_PCM_LINEAR,
channel_layout,
sample_rate,
bits_per_channel,
0);
bytes_per_second_ = audio_parameters_.GetBytesPerSecond();
DCHECK(sink_.get());
if (!is_initialized_) {
sink_->Initialize(
media::SelectSamplesPerPacket(sample_rate),
audio_parameters_.channels,
audio_parameters_.sample_rate,
audio_parameters_.format,
this);
sink_->Start();
is_initialized_ = true;
return true;
}
return false;
}
void AudioRendererImpl::OnStop() {
if (stopped_)
return;
DCHECK(sink_.get());
sink_->Stop();
stopped_ = true;
}
void AudioRendererImpl::SetPlaybackRate(float rate) {
DCHECK_LE(0.0f, rate);
// Handle the case where we stopped due to IO message loop dying.
if (stopped_) {
AudioRendererBase::SetPlaybackRate(rate);
return;
}
// We have two cases here:
// Play: GetPlaybackRate() == 0.0 && rate != 0.0
// Pause: GetPlaybackRate() != 0.0 && rate == 0.0
if (GetPlaybackRate() == 0.0f && rate != 0.0f) {
DoPlay();
} else if (GetPlaybackRate() != 0.0f && rate == 0.0f) {
// Pause is easy, we can always pause.
DoPause();
}
AudioRendererBase::SetPlaybackRate(rate);
}
void AudioRendererImpl::Pause(const base::Closure& callback) {
AudioRendererBase::Pause(callback);
if (stopped_)
return;
DoPause();
}
void AudioRendererImpl::Seek(base::TimeDelta time,
const media::FilterStatusCB& cb) {
AudioRendererBase::Seek(time, cb);
if (stopped_)
return;
DoSeek();
}
void AudioRendererImpl::Play(const base::Closure& callback) {
AudioRendererBase::Play(callback);
if (stopped_)
return;
if (GetPlaybackRate() != 0.0f) {
DoPlay();
} else {
DoPause();
}
}
void AudioRendererImpl::SetVolume(float volume) {
if (stopped_)
return;
DCHECK(sink_.get());
sink_->SetVolume(volume);
}
void AudioRendererImpl::DoPlay() {
earliest_end_time_ = base::Time::Now();
DCHECK(sink_.get());
sink_->Play();
}
void AudioRendererImpl::DoPause() {
DCHECK(sink_.get());
sink_->Pause(false);
}
void AudioRendererImpl::DoSeek() {
earliest_end_time_ = base::Time::Now();
// Pause and flush the stream when we seek to a new location.
DCHECK(sink_.get());
sink_->Pause(true);
}
size_t AudioRendererImpl::Render(const std::vector<float*>& audio_data,
size_t number_of_frames,
size_t audio_delay_milliseconds) {
if (stopped_ || GetPlaybackRate() == 0.0f) {
// Output silence if stopped.
for (size_t i = 0; i < audio_data.size(); ++i)
memset(audio_data[i], 0, sizeof(float) * number_of_frames);
return 0;
}
// Adjust the playback delay.
base::TimeDelta request_delay =
base::TimeDelta::FromMilliseconds(audio_delay_milliseconds);
// Finally we need to adjust the delay according to playback rate.
if (GetPlaybackRate() != 1.0f) {
request_delay = base::TimeDelta::FromMicroseconds(
static_cast<int64>(ceil(request_delay.InMicroseconds() *
GetPlaybackRate())));
}
uint32 bytes_per_frame =
audio_parameters_.bits_per_sample * audio_parameters_.channels / 8;
const size_t buf_size = number_of_frames * bytes_per_frame;
scoped_array<uint8> buf(new uint8[buf_size]);
uint32 filled = FillBuffer(buf.get(), buf_size, request_delay);
DCHECK_LE(filled, buf_size);
UpdateEarliestEndTime(filled, request_delay, base::Time::Now());
uint32 filled_frames = filled / bytes_per_frame;
// Deinterleave each audio channel.
int channels = audio_data.size();
for (int channel_index = 0; channel_index < channels; ++channel_index) {
media::DeinterleaveAudioChannel(buf.get(),
audio_data[channel_index],
channels,
channel_index,
bytes_per_frame / channels,
filled_frames);
// If FillBuffer() didn't give us enough data then zero out the remainder.
if (filled_frames < number_of_frames) {
int frames_to_zero = number_of_frames - filled_frames;
memset(audio_data[channel_index] + filled_frames,
0,
sizeof(float) * frames_to_zero);
}
}
return filled_frames;
}
void AudioRendererImpl::OnRenderError() {
host()->DisableAudioRenderer();
}
void AudioRendererImpl::OnRenderEndOfStream() {
// TODO(enal): schedule callback instead of polling.
if (base::Time::Now() >= earliest_end_time_)
SignalEndOfStream();
}
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