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-rw-r--r--content/renderer/media/audio_track_recorder_unittest.cc4
-rw-r--r--content/renderer/media/media_recorder_handler_unittest.cc2
-rw-r--r--content/renderer/media/renderer_webaudiodevice_impl.cc3
-rw-r--r--content/renderer/media/renderer_webaudiodevice_impl.h4
-rw-r--r--content/renderer/media/webrtc_audio_renderer.cc6
-rw-r--r--content/renderer/media/webrtc_audio_renderer.h4
-rw-r--r--content/renderer/media/webrtc_local_audio_renderer.cc5
-rw-r--r--content/renderer/media/webrtc_local_audio_renderer.h4
8 files changed, 21 insertions, 11 deletions
diff --git a/content/renderer/media/audio_track_recorder_unittest.cc b/content/renderer/media/audio_track_recorder_unittest.cc
index 1966464..f000ea1 100644
--- a/content/renderer/media/audio_track_recorder_unittest.cc
+++ b/content/renderer/media/audio_track_recorder_unittest.cc
@@ -128,7 +128,7 @@ class AudioTrackRecorderTest : public TestWithParam<ATRTestParams> {
audio_track_recorder_->GetOpusBufferDuration(
first_params_.sample_rate()) /
1000));
- first_source_.OnMoreData(bus.get(), 0);
+ first_source_.OnMoreData(bus.get(), 0, 0);
return bus.Pass();
}
scoped_ptr<media::AudioBus> GetSecondSourceAudioBus() {
@@ -138,7 +138,7 @@ class AudioTrackRecorderTest : public TestWithParam<ATRTestParams> {
audio_track_recorder_->GetOpusBufferDuration(
second_params_.sample_rate()) /
1000));
- second_source_.OnMoreData(bus.get(), 0);
+ second_source_.OnMoreData(bus.get(), 0, 0);
return bus.Pass();
}
diff --git a/content/renderer/media/media_recorder_handler_unittest.cc b/content/renderer/media/media_recorder_handler_unittest.cc
index b2a10b9..57fb60d 100644
--- a/content/renderer/media/media_recorder_handler_unittest.cc
+++ b/content/renderer/media/media_recorder_handler_unittest.cc
@@ -110,7 +110,7 @@ class MediaRecorderHandlerTest : public TestWithParam<MediaRecorderTestParams>,
scoped_ptr<media::AudioBus> bus(media::AudioBus::Create(
kTestAudioChannels,
kTestAudioSampleRate * kTestAudioBufferDurationMS / 1000));
- audio_source_.OnMoreData(bus.get(), 0);
+ audio_source_.OnMoreData(bus.get(), 0, 0);
return bus.Pass();
}
diff --git a/content/renderer/media/renderer_webaudiodevice_impl.cc b/content/renderer/media/renderer_webaudiodevice_impl.cc
index d816db3..a0dbd3e 100644
--- a/content/renderer/media/renderer_webaudiodevice_impl.cc
+++ b/content/renderer/media/renderer_webaudiodevice_impl.cc
@@ -94,7 +94,8 @@ double RendererWebAudioDeviceImpl::sampleRate() {
}
int RendererWebAudioDeviceImpl::Render(media::AudioBus* dest,
- int audio_delay_milliseconds) {
+ uint32_t audio_delay_milliseconds,
+ uint32_t frames_skipped) {
#if defined(OS_ANDROID)
if (is_first_buffer_after_silence_) {
DCHECK(!is_using_null_audio_sink_);
diff --git a/content/renderer/media/renderer_webaudiodevice_impl.h b/content/renderer/media/renderer_webaudiodevice_impl.h
index 02c8677..9dd2954 100644
--- a/content/renderer/media/renderer_webaudiodevice_impl.h
+++ b/content/renderer/media/renderer_webaudiodevice_impl.h
@@ -39,7 +39,9 @@ class RendererWebAudioDeviceImpl
double sampleRate() override;
// AudioRendererSink::RenderCallback implementation.
- int Render(media::AudioBus* dest, int audio_delay_milliseconds) override;
+ int Render(media::AudioBus* dest,
+ uint32_t audio_delay_milliseconds,
+ uint32_t frames_skipped) override;
void OnRenderError() override;
diff --git a/content/renderer/media/webrtc_audio_renderer.cc b/content/renderer/media/webrtc_audio_renderer.cc
index a36914a..9714b23 100644
--- a/content/renderer/media/webrtc_audio_renderer.cc
+++ b/content/renderer/media/webrtc_audio_renderer.cc
@@ -418,7 +418,8 @@ media::OutputDeviceStatus WebRtcAudioRenderer::GetDeviceStatus() {
}
int WebRtcAudioRenderer::Render(media::AudioBus* audio_bus,
- int audio_delay_milliseconds) {
+ uint32_t audio_delay_milliseconds,
+ uint32_t frames_skipped) {
DCHECK(audio_renderer_thread_checker_.CalledOnValidThread());
base::AutoLock auto_lock(lock_);
if (!source_)
@@ -427,7 +428,8 @@ int WebRtcAudioRenderer::Render(media::AudioBus* audio_bus,
DVLOG(2) << "WebRtcAudioRenderer::Render()";
DVLOG(2) << "audio_delay_milliseconds: " << audio_delay_milliseconds;
- audio_delay_milliseconds_ = audio_delay_milliseconds;
+ DCHECK_LE(audio_delay_milliseconds, static_cast<uint32_t>(INT_MAX));
+ audio_delay_milliseconds_ = static_cast<int>(audio_delay_milliseconds);
if (audio_fifo_)
audio_fifo_->Consume(audio_bus, audio_bus->frames());
diff --git a/content/renderer/media/webrtc_audio_renderer.h b/content/renderer/media/webrtc_audio_renderer.h
index 0f016eb..1502fb9 100644
--- a/content/renderer/media/webrtc_audio_renderer.h
+++ b/content/renderer/media/webrtc_audio_renderer.h
@@ -168,7 +168,9 @@ class CONTENT_EXPORT WebRtcAudioRenderer
// media::AudioRendererSink::RenderCallback implementation.
// These two methods are called on the AudioOutputDevice worker thread.
- int Render(media::AudioBus* audio_bus, int audio_delay_milliseconds) override;
+ int Render(media::AudioBus* audio_bus,
+ uint32_t audio_delay_milliseconds,
+ uint32_t frames_skipped) override;
void OnRenderError() override;
// Called by AudioPullFifo when more data is necessary.
diff --git a/content/renderer/media/webrtc_local_audio_renderer.cc b/content/renderer/media/webrtc_local_audio_renderer.cc
index 9746e0e..8f0f2ed 100644
--- a/content/renderer/media/webrtc_local_audio_renderer.cc
+++ b/content/renderer/media/webrtc_local_audio_renderer.cc
@@ -32,8 +32,9 @@ enum LocalRendererSinkStates {
} // namespace
// media::AudioRendererSink::RenderCallback implementation
-int WebRtcLocalAudioRenderer::Render(
- media::AudioBus* audio_bus, int audio_delay_milliseconds) {
+int WebRtcLocalAudioRenderer::Render(media::AudioBus* audio_bus,
+ uint32_t audio_delay_milliseconds,
+ uint32_t frames_skipped) {
TRACE_EVENT0("audio", "WebRtcLocalAudioRenderer::Render");
base::AutoLock auto_lock(thread_lock_);
diff --git a/content/renderer/media/webrtc_local_audio_renderer.h b/content/renderer/media/webrtc_local_audio_renderer.h
index 89734c3..6a06984 100644
--- a/content/renderer/media/webrtc_local_audio_renderer.h
+++ b/content/renderer/media/webrtc_local_audio_renderer.h
@@ -97,7 +97,9 @@ class CONTENT_EXPORT WebRtcLocalAudioRenderer
// media::AudioRendererSink::RenderCallback implementation.
// Render() is called on the AudioOutputDevice thread and OnRenderError()
// on the IO thread.
- int Render(media::AudioBus* audio_bus, int audio_delay_milliseconds) override;
+ int Render(media::AudioBus* audio_bus,
+ uint32_t audio_delay_milliseconds,
+ uint32_t frames_skipped) override;
void OnRenderError() override;
// Initializes and starts the |sink_| if