Index: content/renderer/media/webrtc_audio_renderer.cc |
diff --git a/content/renderer/media/webrtc_audio_renderer.cc b/content/renderer/media/webrtc_audio_renderer.cc |
index c21c6624869c6187e18b9d6a46c71e7a7f7874bb..2bf62c479d41db17ddad5212ef07c85938d5458f 100644 |
--- a/content/renderer/media/webrtc_audio_renderer.cc |
+++ b/content/renderer/media/webrtc_audio_renderer.cc |
@@ -8,11 +8,13 @@ |
#include "base/metrics/histogram.h" |
#include "base/string_util.h" |
#include "content/renderer/media/audio_device_factory.h" |
-#include "content/renderer/media/audio_hardware.h" |
+#include "content/renderer/media/renderer_audio_hardware_config.h" |
#include "content/renderer/media/renderer_audio_output_device.h" |
#include "content/renderer/media/webrtc_audio_device_impl.h" |
+#include "content/renderer/render_thread_impl.h" |
#include "media/audio/audio_util.h" |
#include "media/audio/sample_rates.h" |
+ |
#if defined(OS_WIN) |
#include "base/win/windows_version.h" |
#include "media/audio/win/core_audio_util_win.h" |
@@ -24,7 +26,7 @@ namespace { |
// Supported hardware sample rates for output sides. |
#if defined(OS_WIN) || defined(OS_MACOSX) |
-// media::GetAudioOutputHardwareSampleRate() asks the audio layer |
+// RendererAudioHardwareConfig::GetOutputSampleRate() asks the audio layer |
// for its current sample rate (set by the user) on Windows and Mac OS X. |
// The listed rates below adds restrictions and Initialize() |
// will fail if the user selects any rate outside these ranges. |
@@ -106,7 +108,9 @@ bool WebRtcAudioRenderer::Initialize(WebRtcAudioRendererSource* source) { |
// Ask the browser for the default audio output hardware sample-rate. |
// This request is based on a synchronous IPC message. |
henrika (OOO until Aug 14)
2013/01/29 10:46:49
Guess this comment is not 100% any longer since we
DaleCurtis
2013/01/30 01:31:06
Done.
|
- int sample_rate = GetAudioOutputSampleRate(); |
+ RendererAudioHardwareConfig* hardware_config = |
+ RenderThreadImpl::current()->GetAudioHardwareConfig(); |
+ int sample_rate = hardware_config->GetOutputSampleRate(); |
DVLOG(1) << "Audio output hardware sample rate: " << sample_rate; |
UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioOutputSampleRate", |
sample_rate, media::kUnexpectedAudioSampleRate); |