Added an "enable-audio-processor" flag and WebRtcAudioProcessor class

content/renderer/media/webrtc_audio_processor.h

+// Copyright 2013 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.
+
+#ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_PROCESSOR_H_
+#define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_PROCESSOR_H_
+
+#include "base/atomicops.h"
+#include "base/synchronization/lock.h"
+#include "base/threading/thread_checker.h"
+#include "base/time/time.h"
+#include "content/common/content_export.h"
+#include "media/base/audio_converter.h"
+#include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h"
+#include "third_party/webrtc/modules/audio_processing/include/audio_processing.h"
+#include "third_party/webrtc/modules/interface/module_common_types.h"
+
+namespace media {
+class AudioBus;
+class AudioFifo;
+class AudioParameters;
+}  // namespace media
+
+namespace webrtc {
+class AudioFrame;
+}
+
+namespace content {
+
+// This class owns an object of webrtc::AudioProcessing which contains signal
+// processing components like AGC, AEC and NS. It enables the components based
+// on the getUserMedia constraints, processes the data and outputs it in a unit
+// of 10 ms data chunk.
+class CONTENT_EXPORT WebRtcAudioProcessor {
+ public:
+  explicit WebRtcAudioProcessor(
+      const webrtc::MediaConstraintsInterface* constraints);
+  ~WebRtcAudioProcessor();
+
+  // Pushes capture data in |audio_source| to the internal FIFO.
+  // Called on the capture audio thread.
+  void PushCaptureData(media::AudioBus* audio_source);
+
+  // Push the render audio to WebRtc::AudioProcessing for analysis. This is

[tommi (sloooow) - chröme, 2013/11/21 19:52:03]

Since AudioTrack isn't strictly speaking WebRTC specific (assuming webrtc means
PeerConnection), then I'm not sure that AudioProcessing should be in a webrtc
namespace or have WebRTC as part of its name.

[no longer working on chromium, 2013/11/22 13:27:53]

Changed it to webrtc::AudioProcessing, which is the name of the class with
webrtc namespace.

+  // needed iff echo processing is enabled.
+  // |render_audio| is the pointer to the render audio data, its format
+  // is specified by |sample_rate|, |number_of_channels| and |number_of_frames|.
+  // Called on the render audio thread.
+  void PushRenderData(const int16* render_audio,
+                      int sample_rate,
+                      int number_of_channels,
+                      int number_of_frames,
+                      base::TimeDelta render_delay);
+
+  // Processes a block of 10 ms data from the internal FIFO and outputs it via
+  // |out|. |out| is the address of the pointer that will be pointed to
+  // the post-processed data if the method is returning a true. The lifetime
+  // of the data represeted by |out| is guaranteed to outlive the method call.

[Henrik Grunell, 2013/11/22 09:03:36]

Are there any guarantees for how long the data pointed to by *|out| will live
*after* this function has returned? Is the caller supposed to copy the data? In
that case, it seems better to copy the data in this function to memory the
caller has allocated.

[no longer working on chromium, 2013/11/22 13:27:53]

As explained in the comment, *|out| guarantees to outlive the method call, so
the caller is supposed to get the pointer of the data, then demux it to all the
sinks, and the sink will copy the data.

We are trying to avoid an extra memcopy here to help performance.

[Henrik Grunell, 2013/11/25 10:52:02]

We talked about this offline too. I don't think the comment is clear, hence my
original question. I think it should be clarified that the data pointed to by
*|out| won't change until this function is called again. (If that's correct?)

[no longer working on chromium, 2013/11/25 16:46:35]

I missed this one before, now it is done.

+  // Returns true if the internal FIFO has at least 10 ms data for processing,
+  // otherwise false.
+  // |capture_delay|, |volume| and |key_pressed| will be passed to
+  // webrtc::AudioProcessing to help processing the data.
+  // Called on the capture audio thread.
+  bool ProcessAndConsumeData(base::TimeDelta capture_delay,
+                             int volume,
+                             bool key_pressed,
+                             int16** out);
+
+  // Called when the format of the capture data has changed.
+  // This has to be called before PushCaptureData() and ProcessAndConsumeData().
+  // Called on the main render thread.
+  void SetCaptureFormat(const media::AudioParameters& source_params);
+
+  // The audio format of the output from the processor.
+  const media::AudioParameters& OutputFormat() const;
+
+  // Accessor to check if the audio processing is enabled or not.
+  bool has_audio_processing() const { return audio_processing_.get() != NULL; }
+
+ private:
+  class WebRtcAudioConverter;
+
+  // Helper to initialize the WebRtc AudioProcessing.
+  void InitializeAudioProcessingModule(
+      const webrtc::MediaConstraintsInterface* constraints);
+
+  // Helper to initialize the render converter.
+  void InitializeRenderConverterIfNeeded(int sample_rate,
+                                         int number_of_channels,
+                                         int frames_per_buffer);
+
+  // Called by ProcessAndConsumeData().
+  void ProcessData(webrtc::AudioFrame* audio_frame,
+                   base::TimeDelta capture_delay,
+                   int volume,
+                   bool key_pressed);
+
+  // Called when the processor is going away.
+  void StopAudioProcessing();
+
+  // Cached value for the render delay latency.
+  base::subtle::Atomic32 render_delay_ms_;

[tommi (sloooow) - chröme, 2013/11/21 19:52:03]

can you add some comments on why it's an Atomic32 value?
the comments are generally good here btw.

[no longer working on chromium, 2013/11/22 13:27:53]

Done.

+
+  // WebRtc AudioProcessing module which does AEC, AGC, NS, HighPass filter,
+  // ..etc.
+  scoped_ptr<webrtc::AudioProcessing> audio_processing_;
+
+  // Converter used for the down-mixing and resampling of the capture data.
+  scoped_ptr<WebRtcAudioConverter> capture_converter_;
+
+  // AudioFrame used to hold the output of |capture_converter_|.
+  webrtc::AudioFrame capture_frame_;
+
+  // Converter used for the down-mixing and resampling of the render data when
+  // the AEC is enabled.
+  scoped_ptr<WebRtcAudioConverter> render_converter_;
+
+  // AudioFrame used to hold the output of |render_converter_|.
+  webrtc::AudioFrame render_frame_;
+
+  // Data bus to help converting interleaved data to an AudioBus.
+  scoped_ptr<media::AudioBus> render_data_bus_;
+
+  // Used to DCHECK that some methods are called on the main render thread.
+  base::ThreadChecker main_thread_checker_;
+
+  // Used to DCHECK that some methods are called on the capture audio thread.
+  base::ThreadChecker capture_thread_checker_;
+
+  // Used to DCHECK that PushRenderData() is called on the render audio thread.
+  base::ThreadChecker render_thread_checker_;
+};
+
+}  // namespace content
+
+#endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_PROCESSOR_H_