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Side by Side Diff: content/renderer/media/webrtc_local_audio_track.h

Issue 37793005: move the APM to chrome. (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: Created 7 years, 2 months ago
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1 // Copyright 2013 The Chromium Authors. All rights reserved. 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 2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file. 3 // found in the LICENSE file.
4 4
5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ 5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_
6 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_
7 7
8 #include <list> 8 #include <list>
9 #include <string> 9 #include <string>
10 10
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58 58
59 // Starts the local audio track. Called on the main render thread and 59 // Starts the local audio track. Called on the main render thread and
60 // should be called only once when audio track is created. 60 // should be called only once when audio track is created.
61 void Start(); 61 void Start();
62 62
63 // Stops the local audio track. Called on the main render thread and 63 // Stops the local audio track. Called on the main render thread and
64 // should be called only once when audio track going away. 64 // should be called only once when audio track going away.
65 void Stop(); 65 void Stop();
66 66
67 // Method called by the capturer to deliver the capture data. 67 // Method called by the capturer to deliver the capture data.
68 void Capture(media::AudioBus* audio_source, 68 void Capture(const int16* audio_data,
69 int audio_delay_milliseconds, 69 int sample_rate,
70 int volume, 70 int number_of_channels,
71 bool key_pressed); 71 int number_of_frames);
72 72
73 // Method called by the capturer to set the audio parameters used by source 73 // Method called by the capturer to set the audio parameters used by source
74 // of the capture data.. 74 // of the capture data..
75 // Can be called on different user threads. 75 // Can be called on different user threads.
76 void SetCaptureFormat(const media::AudioParameters& params); 76 void SetCaptureFormat(const media::AudioParameters& params);
77 77
78 protected: 78 protected:
79 WebRtcLocalAudioTrack( 79 WebRtcLocalAudioTrack(
80 const std::string& label, 80 const std::string& label,
81 const scoped_refptr<WebRtcAudioCapturer>& capturer, 81 const scoped_refptr<WebRtcAudioCapturer>& capturer,
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118 base::ThreadChecker thread_checker_; 118 base::ThreadChecker thread_checker_;
119 119
120 // Protects |params_| and |sinks_|. 120 // Protects |params_| and |sinks_|.
121 mutable base::Lock lock_; 121 mutable base::Lock lock_;
122 122
123 // A vector of WebRtc VoE channels that the capturer sends datat to. 123 // A vector of WebRtc VoE channels that the capturer sends datat to.
124 std::vector<int> voe_channels_; 124 std::vector<int> voe_channels_;
125 125
126 bool need_audio_processing_; 126 bool need_audio_processing_;
127 127
128 // Buffers used for temporary storage during capture callbacks. 128 media::AudioParameters params_;
129 // Allocated during initialization.
130 class ConfiguredBuffer;
131 scoped_refptr<ConfiguredBuffer> buffer_;
132 129
133 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack); 130 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack);
134 }; 131 };
135 132
136 } // namespace content 133 } // namespace content
137 134
138 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ 135 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_
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