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1 // Copyright 2013 The Chromium Authors. All rights reserved. | |
2 // Use of this source code is governed by a BSD-style license that can be | |
3 // found in the LICENSE file. | |
4 | |
5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_PROCESSOR_H_ | |
6 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_PROCESSOR_H_ | |
7 | |
8 #include "base/synchronization/lock.h" | |
9 #include "content/common/content_export.h" | |
10 #include "media/base/audio_converter.h" | |
11 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface .h" | |
12 #include "third_party/webrtc/modules/audio_processing/include/audio_processing.h " | |
13 #include "third_party/webrtc/modules/interface/module_common_types.h" | |
14 | |
15 namespace media { | |
16 class AudioBus; | |
17 class AudioFifo; | |
18 class AudioParameters; | |
19 } // namespace media | |
20 | |
21 namespace webrtc { | |
22 class AudioFrame; | |
23 } | |
24 | |
25 namespace content { | |
26 | |
27 // This class is a wrapper class of webrtc::AudioProcessing. | |
28 class CONTENT_EXPORT WebRtcAudioProcessor { | |
29 public: | |
30 explicit WebRtcAudioProcessor( | |
31 const webrtc::MediaConstraintsInterface* constraints); | |
32 ~WebRtcAudioProcessor(); | |
33 | |
34 // TODO(xians): Add comment. | |
35 void SetFormat(const media::AudioParameters& source_params); | |
36 | |
37 void Push(media::AudioBus* audio_source); | |
Henrik Grunell
2013/10/31 11:56:12
Add comment.
| |
38 | |
39 // Returns true if it has 10ms data for processing, otherwise false. | |
40 bool ProcessAndConsume10MsData(int capture_audio_delay_ms, | |
41 int volume, | |
42 bool key_pressed); | |
43 | |
44 const int16* OutputBuffer() const; | |
45 const media::AudioParameters& OutputFormat() const; | |
46 | |
47 // Feed render audio to AudioProcessing for analysis. This is needed | |
48 // if and only if echo processing is enabled. | |
49 void FeedRenderDataToAudioProcessing(const int16* render_audio, | |
50 int sample_rate, | |
51 int number_of_channels, | |
52 int number_of_frames, | |
53 int render_delay_ms); | |
54 | |
55 bool has_audio_processing() const { return audio_processing_.get() != NULL; } | |
56 | |
57 private: | |
58 class WebRtcAudioConverter; | |
59 | |
60 void InitializeAudioProcessingModule( | |
61 const webrtc::MediaConstraintsInterface* constraints); | |
62 void InitializeRenderConverterIfNeeded(int sample_rate, | |
63 int number_of_channels, | |
64 int frames_per_buffer); | |
65 // Processes 10ms data. | |
66 void Process10MsData(int audio_delay_milliseconds, | |
67 int volume, | |
68 bool key_pressed); | |
69 | |
70 void StopAudioProcessing(); | |
71 | |
72 // Cached value for the render delay latency. | |
73 int render_delay_ms_; | |
74 | |
75 // Protects |render_delay_ms_|. | |
76 // TODO(xians): Can we get rid of the lock? | |
77 mutable base::Lock lock_; | |
78 | |
79 // Hanles processing the audio data. | |
80 scoped_ptr<webrtc::AudioProcessing> audio_processing_; | |
81 | |
82 scoped_ptr<WebRtcAudioConverter> capture_converter_; | |
83 scoped_ptr<WebRtcAudioConverter> render_converter_; | |
84 }; | |
85 | |
86 } // namespace content | |
87 | |
88 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_PROCESSOR_H_ | |
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