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Unified Diff: content/renderer/media/webrtc_audio_device_impl.h

Issue 1834323002: MediaStream audio: Refactor 3 separate "glue" implementations into one. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Comments about PS1 addressed. Fixed is_stopped_/StopSource() foo. REBASE Created 4 years, 9 months ago
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Index: content/renderer/media/webrtc_audio_device_impl.h
diff --git a/content/renderer/media/webrtc_audio_device_impl.h b/content/renderer/media/webrtc_audio_device_impl.h
index de92afe4dda4d47ef36421178eecf030521cd41d..faea56d548fc8f09ec22d0a878ff66d8c70cddd6 100644
--- a/content/renderer/media/webrtc_audio_device_impl.h
+++ b/content/renderer/media/webrtc_audio_device_impl.h
@@ -19,11 +19,9 @@
#include "base/memory/scoped_ptr.h"
#include "base/threading/thread_checker.h"
#include "content/common/content_export.h"
-#include "content/renderer/media/webrtc_audio_capturer.h"
+#include "content/renderer/media/webrtc/processed_local_audio_source.h"
#include "content/renderer/media/webrtc_audio_device_not_impl.h"
#include "ipc/ipc_platform_file.h"
-#include "media/base/audio_capturer_source.h"
-#include "media/base/audio_renderer_sink.h"
// A WebRtcAudioDeviceImpl instance implements the abstract interface
// webrtc::AudioDeviceModule which makes it possible for a user (e.g. webrtc::
@@ -184,7 +182,7 @@
namespace content {
-class WebRtcAudioCapturer;
+class ProcessedLocalAudioSource;
class WebRtcAudioRenderer;
// TODO(xians): Move the following two interfaces to webrtc so that
@@ -311,8 +309,8 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
// Capturers must remain valid until RemoveAudioCapturer() is called.
// TODO(xians): Remove these two methods once the ADM does not need to pass
// hardware information up to WebRtc.
- void AddAudioCapturer(WebRtcAudioCapturer* capturer);
- void RemoveAudioCapturer(WebRtcAudioCapturer* capturer);
+ void AddAudioCapturer(ProcessedLocalAudioSource* capturer);
+ void RemoveAudioCapturer(ProcessedLocalAudioSource* capturer);
// Gets paired device information of the capture device for the audio
// renderer. This is used to pass on a session id, sample rate and buffer
@@ -329,7 +327,7 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
}
private:
- typedef std::list<WebRtcAudioCapturer*> CapturerList;
+ typedef std::list<ProcessedLocalAudioSource*> CapturerList;
typedef std::list<WebRtcPlayoutDataSource::Sink*> PlayoutDataSinkList;
class RenderBuffer;

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