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

Issue 1966043006: Revert of MediaStream audio: Refactor 3 separate "glue" implementations into one. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 4 years, 7 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 6af4a86935616b627b7874177142486df6638100..52cb2a0dd5b34b78e5eb92508ce15ce05f19b36c 100644
--- a/content/renderer/media/webrtc_audio_device_impl.h
+++ b/content/renderer/media/webrtc_audio_device_impl.h
@@ -19,8 +19,11 @@
#include "base/memory/ref_counted.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_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::
@@ -179,13 +182,9 @@
// transferring maximum levels between the renderer and the browser.
//
-namespace media {
-class AudioBus;
-}
-
namespace content {
-class ProcessedLocalAudioSource;
+class WebRtcAudioCapturer;
class WebRtcAudioRenderer;
// TODO(xians): Move the following two interfaces to webrtc so that
@@ -312,8 +311,8 @@
// 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(ProcessedLocalAudioSource* capturer);
- void RemoveAudioCapturer(ProcessedLocalAudioSource* capturer);
+ void AddAudioCapturer(WebRtcAudioCapturer* capturer);
+ void RemoveAudioCapturer(WebRtcAudioCapturer* 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
@@ -330,7 +329,7 @@
}
private:
- typedef std::list<ProcessedLocalAudioSource*> CapturerList;
+ typedef std::list<WebRtcAudioCapturer*> CapturerList;
typedef std::list<WebRtcPlayoutDataSource::Sink*> PlayoutDataSinkList;
class RenderBuffer;
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