Index: content/renderer/media/audio_device.cc |
=================================================================== |
--- content/renderer/media/audio_device.cc (revision 107169) |
+++ content/renderer/media/audio_device.cc (working copy) |
@@ -14,6 +14,19 @@ |
#include "content/renderer/render_thread_impl.h" |
#include "media/audio/audio_util.h" |
+AudioDevice::AudioDevice(RenderCallback* callback) |
+ : buffer_size_(0), |
scherkus (not reviewing)
2011/11/09 02:39:05
indent by two more
Chris Rogers
2011/11/10 02:17:22
Done.
|
+ channels_(0), |
+ bits_per_sample_(16), |
+ sample_rate_(0), |
+ latency_format_(AudioParameters::AUDIO_PCM_LOW_LATENCY), |
+ callback_(callback), |
+ audio_delay_milliseconds_(0), |
+ volume_(1.0), |
+ stream_id_(0) { |
+ filter_ = RenderThreadImpl::current()->audio_message_filter(); |
+} |
+ |
AudioDevice::AudioDevice(size_t buffer_size, |
int channels, |
double sample_rate, |
@@ -27,6 +40,29 @@ |
volume_(1.0), |
stream_id_(0) { |
filter_ = RenderThreadImpl::current()->audio_message_filter(); |
+ Initialize(buffer_size, |
+ channels, |
+ sample_rate, |
+ AudioParameters::AUDIO_PCM_LOW_LATENCY); |
+} |
+ |
+void AudioDevice::Initialize(size_t buffer_size, |
+ int channels, |
+ double sample_rate, |
+ AudioParameters::Format latency_format) { |
+ CHECK_EQ(0, stream_id_); |
scherkus (not reviewing)
2011/11/09 02:39:05
nit: perhaps add logging?
CHECK_EQ(0, stream_id) <
Chris Rogers
2011/11/10 02:17:22
Done.
|
+ if (stream_id_) |
+ return; |
+ |
+ buffer_size_ = buffer_size; |
+ channels_ = channels; |
+ sample_rate_ = sample_rate; |
+ latency_format_ = latency_format; |
+ |
+ // Cleanup from any previous initialization. |
+ for (size_t i = 0; i < audio_data_.size(); ++i) |
+ delete [] audio_data_[i]; |
acolwell GONE FROM CHROMIUM
2011/11/08 21:44:20
How about changing audio_data_ to use scoped_array
Chris Rogers
2011/11/10 02:17:22
It seems that scoped_array<> cannot be used with S
|
+ |
audio_data_.reserve(channels); |
for (int i = 0; i < channels; ++i) { |
float* channel_data = new float[buffer_size]; |
@@ -34,6 +70,10 @@ |
} |
} |
+bool AudioDevice::IsInitialized() { |
+ return audio_data_.size() > 0; |
+} |
+ |
AudioDevice::~AudioDevice() { |
// The current design requires that the user calls Stop() before deleting |
// this class. |
@@ -43,8 +83,11 @@ |
} |
void AudioDevice::Start() { |
+ if (stream_id_) |
acolwell GONE FROM CHROMIUM
2011/11/08 21:44:20
Should this be a CHECK_EQ() like elsewhere? We wan
henrika (OOO until Aug 14)
2011/11/09 17:05:22
Why is the existing check in InitializeOnIOThread(
Chris Rogers
2011/11/10 02:17:22
I believe Henrik is right here. This check is not
|
+ return; |
+ |
AudioParameters params; |
- params.format = AudioParameters::AUDIO_PCM_LOW_LATENCY; |
+ params.format = latency_format_; |
params.channels = channels_; |
params.sample_rate = static_cast<int>(sample_rate_); |
params.bits_per_sample = bits_per_sample_; |
@@ -56,6 +99,9 @@ |
} |
bool AudioDevice::Stop() { |
+ if (!stream_id_) |
acolwell GONE FROM CHROMIUM
2011/11/08 21:44:20
CHECK_NE(0, stream_id_) ?
henrika (OOO until Aug 14)
2011/11/09 17:05:22
Similar comment as for Start().
Chris Rogers
2011/11/10 02:17:22
Yes the check that's already in ShutDownOnIOThread
|
+ return true; |
+ |
// Max waiting time for Stop() to complete. If this time limit is passed, |
// we will stop waiting and return false. It ensures that Stop() can't block |
// the calling thread forever. |
@@ -84,6 +130,18 @@ |
return true; |
} |
+void AudioDevice::Play() { |
+ ChildProcess::current()->io_message_loop()->PostTask( |
+ FROM_HERE, |
+ base::Bind(&AudioDevice::PlayOnIOThread, this)); |
+} |
+ |
+void AudioDevice::Pause(bool flush) { |
henrika (OOO until Aug 14)
2011/11/09 17:05:22
What happens if a user calls Start(), Pause(), Pla
Chris Rogers
2011/11/10 02:17:22
Yes, I think we can make it even simpler and consi
henrika (OOO until Aug 14)
2011/11/10 11:45:07
Fine by me. In general I prefer simple and readabl
Chris Rogers
2011/11/15 22:48:29
I think I've addressed this problem with |play_on_
|
+ ChildProcess::current()->io_message_loop()->PostTask( |
+ FROM_HERE, |
+ base::Bind(&AudioDevice::PauseOnIOThread, this, flush)); |
+} |
+ |
bool AudioDevice::SetVolume(double volume) { |
if (volume < 0 || volume > 1.0) |
return false; |
@@ -103,7 +161,7 @@ |
} |
void AudioDevice::InitializeOnIOThread(const AudioParameters& params) { |
- // Make sure we don't call Start() more than once. |
+ // Make sure we don't create the stream more than once. |
DCHECK_EQ(0, stream_id_); |
if (stream_id_) |
return; |
@@ -112,11 +170,19 @@ |
Send(new AudioHostMsg_CreateStream(stream_id_, params, true)); |
} |
-void AudioDevice::StartOnIOThread() { |
+void AudioDevice::PlayOnIOThread() { |
if (stream_id_) |
Send(new AudioHostMsg_PlayStream(stream_id_)); |
} |
+void AudioDevice::PauseOnIOThread(bool flush) { |
+ if (stream_id_) { |
+ Send(new AudioHostMsg_PauseStream(stream_id_)); |
+ if (flush) |
+ Send(new AudioHostMsg_FlushStream(stream_id_)); |
+ } |
+} |
+ |
void AudioDevice::ShutDownOnIOThread(base::WaitableEvent* completion) { |
// Make sure we don't call shutdown more than once. |
if (!stream_id_) { |
@@ -138,20 +204,17 @@ |
void AudioDevice::OnRequestPacket(AudioBuffersState buffers_state) { |
// This method does not apply to the low-latency system. |
- NOTIMPLEMENTED(); |
acolwell GONE FROM CHROMIUM
2011/11/08 21:44:20
Why? This and the 2 below.
Chris Rogers
2011/11/10 02:17:22
These are unrelated changes - just part of general
|
} |
void AudioDevice::OnStateChanged(AudioStreamState state) { |
if (state == kAudioStreamError) { |
DLOG(WARNING) << "AudioDevice::OnStateChanged(kError)"; |
} |
- NOTIMPLEMENTED(); |
} |
void AudioDevice::OnCreated( |
base::SharedMemoryHandle handle, uint32 length) { |
// Not needed in this simple implementation. |
- NOTIMPLEMENTED(); |
} |
void AudioDevice::OnLowLatencyCreated( |
@@ -191,7 +254,7 @@ |
MessageLoop::current()->PostTask( |
FROM_HERE, |
- base::Bind(&AudioDevice::StartOnIOThread, this)); |
+ base::Bind(&AudioDevice::PlayOnIOThread, this)); |
} |
void AudioDevice::OnVolume(double volume) { |
@@ -210,10 +273,8 @@ |
const int samples_per_ms = static_cast<int>(sample_rate_) / 1000; |
const int bytes_per_ms = channels_ * (bits_per_sample_ / 8) * samples_per_ms; |
- while ((sizeof(pending_data) == socket_->Receive(&pending_data, |
- sizeof(pending_data))) && |
- (pending_data >= 0)) { |
acolwell GONE FROM CHROMIUM
2011/11/08 21:44:20
Why is it safe to remove the >= 0 check?
Chris Rogers
2011/11/10 02:17:22
We need to remove this check, because this case co
acolwell GONE FROM CHROMIUM
2011/11/10 21:58:15
Won't allow pending_data < 0 break the audio_delay
Chris Rogers
2011/11/15 22:48:29
Yes, good point -- fixed.
On 2011/11/10 21:58:15,
|
- |
+ while (sizeof(pending_data) == |
+ socket_->Receive(&pending_data, sizeof(pending_data))) { |
// Convert the number of pending bytes in the render buffer |
// into milliseconds. |
audio_delay_milliseconds_ = pending_data / bytes_per_ms; |