Index: content/renderer/pepper/pepper_media_stream_audio_track_host.cc |
diff --git a/content/renderer/pepper/pepper_media_stream_audio_track_host.cc b/content/renderer/pepper/pepper_media_stream_audio_track_host.cc |
new file mode 100644 |
index 0000000000000000000000000000000000000000..752db08ad8caae4abe0be7d8cafb90b599caeff9 |
--- /dev/null |
+++ b/content/renderer/pepper/pepper_media_stream_audio_track_host.cc |
@@ -0,0 +1,175 @@ |
+// Copyright (c) 2014 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/pepper/pepper_media_stream_audio_track_host.h" |
+ |
+#include "base/bind.h" |
+#include "base/location.h" |
+#include "base/logging.h" |
+#include "base/macros.h" |
+#include "base/message_loop/message_loop_proxy.h" |
+#include "ppapi/c/pp_errors.h" |
+#include "ppapi/c/ppb_audio_frame.h" |
+#include "ppapi/shared_impl/media_stream_frame.h" |
+ |
+using media::AudioParameters; |
+ |
+namespace { |
+ |
+// TODO(penghuang): make it configurable. |
+const int32_t kNumberOfFrames = 4; |
+ |
+} // namespace |
+ |
+namespace content { |
+ |
+PepperMediaStreamAudioTrackHost::AudioSink::AudioSink( |
+ PepperMediaStreamAudioTrackHost* host) |
+ : host_(host), |
+ frame_data_size_(0), |
+ frames_(0), |
+ main_message_loop_proxy_(base::MessageLoopProxy::current()) { |
+} |
+ |
+PepperMediaStreamAudioTrackHost::AudioSink::~AudioSink() {} |
+ |
+void PepperMediaStreamAudioTrackHost::AudioSink::EnqueueFrame(int32_t index) { |
+ DCHECK_GE(index, 0); |
yzshen1
2014/01/29 21:28:06
Please also also DCHECK that this is called on the
Peng
2014/01/31 18:54:43
Done.
|
+ DCHECK_LT(index, host_->frame_buffer()->number_of_frames()); |
+ DCHECK(!(base::subtle::NoBarrier_Load(&frames_) & (1 << index))); |
+ |
+ // Set the bit for the frame index. Automic does not support and/or |
+ // operations, so we use add/sub instead. |
+ base::subtle::NoBarrier_AtomicIncrement(&frames_, 1 << index); |
yzshen1
2014/01/29 21:28:06
This kind of code is, as the name suggested, subtl
Peng
2014/01/31 18:54:43
Atomic's performance is slightly better than lock.
yzshen1
2014/01/31 19:50:26
IMO, the biggest reason that we avoid lock is to m
dmichael (off chromium)
2014/01/31 20:14:28
Yuzhu's exactly right; I was only concerned about
Peng
2014/01/31 21:09:25
Done.
Peng
2014/01/31 21:09:25
Fixed by using base::Lock. Done
|
+} |
+ |
+void PepperMediaStreamAudioTrackHost::AudioSink::InitFramesOnMainThread( |
+ uint32_t number_of_frames, uint32_t frame_size) { |
+ // Maybe use Atomic64 to support more frames. |
+ DCHECK_LE(number_of_frames, 32u); |
+ bool result = host_->InitFrames(number_of_frames, frame_size); |
+ DCHECK(result); |
+ for (uint32_t i = 0; i < number_of_frames; ++i) { |
+ int32_t index = host_->frame_buffer()->DequeueFrame(); |
+ DCHECK_GE(index, 0); |
+ } |
+ |
+ base::subtle::Atomic32 frames = base::subtle::NoBarrier_CompareAndSwap( |
+ &frames_, 0, (1 << number_of_frames) - 1); |
+ DCHECK(!frames); |
+} |
+ |
+void PepperMediaStreamAudioTrackHost::AudioSink::OnData(const int16* audio_data, |
+ int sample_rate, |
+ int number_of_channels, |
+ int number_of_frames) { |
+ DCHECK(audio_data); |
+ DCHECK_EQ(sample_rate, audio_params_.sample_rate()); |
+ DCHECK_EQ(number_of_channels, audio_params_.channels()); |
+ DCHECK_EQ(number_of_frames, audio_params_.frames_per_buffer()); |
+ |
+ base::subtle::Atomic32 frames = base::subtle::NoBarrier_Load(&frames_); |
+ if (frames) { |
+ // If frames isn't zero, the |frame_buffer()| should be initialized already, |
+ // and |frame_buffer()|'s attributes (|number_of_frames()|, |frame_size()|, |
+ // etc) will not be changed. So it is safe to read them in the audio thread. |
+ int32_t index = 0; |
+ int32_t n = host_->frame_buffer()->number_of_frames(); |
+ |
+ // Find a free frame. |
+ while(index < n && !(frames & (1 << index))) |
+ index ++; |
yzshen1
2014/01/29 21:28:06
no space in the middle, please.
Peng
2014/01/31 18:54:43
Done.
|
+ |
+ if (index < n) { |
+ // TODO(penghuang): support re-sampling, etc. |
+ ppapi::MediaStreamFrame::Audio* frame = |
+ &(host_->frame_buffer()->GetFramePointer(index)->audio); |
yzshen1
2014/01/29 21:28:06
We are accessing the frame buffer from multiple th
Peng
2014/01/31 18:54:43
Yes. After the frame_buffer() being initialized, t
|
+ frame->header.size = host_->frame_buffer()->frame_size(); |
+ frame->header.type = ppapi::MediaStreamFrame::TYPE_AUDIO; |
+ frame->timestamp = timestamp_.InMillisecondsF(); |
+ frame->sample_rate = static_cast<PP_AudioFrame_SampleRate>(sample_rate); |
+ frame->number_of_channels = number_of_channels; |
+ frame->number_of_samples = number_of_channels * number_of_frames; |
+ frame->data_size = frame_data_size_; |
+ memcpy(frame->data, audio_data, frame_data_size_); |
+ |
+ // Clear the bit for the frame index. Automic does not support and/or |
+ // operations, so we use add/sub instead. |
+ base::subtle::NoBarrier_AtomicIncrement(&frames_, -(1 << index)); |
yzshen1
2014/01/29 21:28:06
what if index == 31?
Peng
2014/01/31 18:54:43
I think (1 << 32) will be 0x80000000. Because it i
|
+ |
+ // This function is called from the audio thread, but |
+ // |SendEnqueueFrameMessageToPlugin()| doesn't use any sync IPC, |
+ // so it is safe to call it from the audio thread directly. |
+ host_->SendEnqueueFrameMessageToPlugin(index); |
yzshen1
2014/01/29 21:28:06
Unless we said explicitly comment at the declarati
Peng
2014/01/31 18:54:43
I understand. But I really don't have any good ide
|
+ } |
+ } |
+ timestamp_ += frame_duration_; |
+} |
+ |
+void PepperMediaStreamAudioTrackHost::AudioSink::OnSetFormat( |
+ const AudioParameters& params) { |
+ DCHECK(!audio_params_.IsValid()); |
+ DCHECK(params.IsValid()); |
+ DCHECK_EQ(params.bits_per_sample(), 16); |
+ DCHECK((params.sample_rate() == AudioParameters::kTelephoneSampleRate) || |
+ (params.sample_rate() == AudioParameters::kAudioCDSampleRate)); |
+ |
+ COMPILE_ASSERT(AudioParameters::kTelephoneSampleRate == 8000, |
+ audio_sample_rate_does_not_match); |
+ COMPILE_ASSERT(AudioParameters::kAudioCDSampleRate == 44100, |
+ audio_sample_rate_does_not_match); |
+ |
+ // TODO(penghuang): support setting format more than once. |
+ audio_params_ = params; |
+ frame_duration_ = audio_params_.GetBufferDuration(); |
+ frame_data_size_ = audio_params_.GetBytesPerBuffer(); |
+ |
+ // The size is slightly bigger than necessary, because 8 extra bytes are added |
+ // into the struct. Also see |MediaStreamFrame|. |
+ int32_t size = sizeof(ppapi::MediaStreamFrame::Audio) + frame_data_size_; |
+ |
+ // This function is called from the audio thread and |InitFrames()| uses some |
+ // sync IPC. So we have to call |InitFrames()| from the main thread. |
yzshen1
2014/01/29 21:28:06
I have no problem with running InitFrames() on the
Peng
2014/01/31 18:54:43
Currently, InitFrames() uses content::RenderThread
yzshen1
2014/02/03 18:14:36
Ah, I see. :)
I thought you meant that you added
|
+ main_message_loop_proxy_->PostTask( |
+ FROM_HERE, |
+ base::Bind(&AudioSink::InitFramesOnMainThread, |
+ AsWeakPtr(), kNumberOfFrames, size)); |
+} |
+ |
+PepperMediaStreamAudioTrackHost::PepperMediaStreamAudioTrackHost( |
+ RendererPpapiHost* host, |
+ PP_Instance instance, |
+ PP_Resource resource, |
+ const blink::WebMediaStreamTrack& track) |
+ : PepperMediaStreamTrackHostBase(host, instance, resource), |
+ track_(track), |
+ connected_(false), |
+ audio_sink_(this) { |
+ DCHECK(!track_.isNull()); |
+} |
+ |
+PepperMediaStreamAudioTrackHost::~PepperMediaStreamAudioTrackHost() { |
+ OnClose(); |
yzshen1
2014/01/29 21:28:06
(I haven't read the relevant backend code, but wan
Peng
2014/01/31 18:54:43
https://code.google.com/p/chromium/codesearch#chro
|
+} |
+ |
+void PepperMediaStreamAudioTrackHost::OnClose() { |
+ if (connected_) { |
+ MediaStreamAudioSink::RemoveFromAudioTrack(&audio_sink_, track_); |
+ connected_ = false; |
+ } |
+} |
+ |
+bool PepperMediaStreamAudioTrackHost::OnNewFramePreEnqueued(int32_t index) { |
+ audio_sink_.EnqueueFrame(index); |
+ return true; |
+} |
+ |
+void PepperMediaStreamAudioTrackHost::DidConnectPendingHostToResource() { |
+ if (!connected_) { |
+ MediaStreamAudioSink::AddToAudioTrack(&audio_sink_, track_); |
+ connected_ = true; |
+ } |
+} |
+ |
+} // namespace content |