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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_SOURCE_PROVIDER_H_ | 5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_SOURCE_PROVIDER_H_ |
| 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_SOURCE_PROVIDER_H_ | 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_SOURCE_PROVIDER_H_ |
| 7 | 7 |
| 8 #include <stddef.h> | 8 #include <stddef.h> |
| 9 | 9 |
| 10 #include <memory> | 10 #include <memory> |
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| 68 | 68 |
| 69 // blink::WebAudioSourceProvider implementation. | 69 // blink::WebAudioSourceProvider implementation. |
| 70 void setClient(blink::WebAudioSourceProviderClient* client) override; | 70 void setClient(blink::WebAudioSourceProviderClient* client) override; |
| 71 void provideInput(const blink::WebVector<float*>& audio_data, | 71 void provideInput(const blink::WebVector<float*>& audio_data, |
| 72 size_t number_of_frames) override; | 72 size_t number_of_frames) override; |
| 73 | 73 |
| 74 // media::AudioConverter::Inputcallback implementation. | 74 // media::AudioConverter::Inputcallback implementation. |
| 75 // This function is triggered by provideInput()on the WebAudio audio thread, | 75 // This function is triggered by provideInput()on the WebAudio audio thread, |
| 76 // so it has been under the protection of |lock_|. | 76 // so it has been under the protection of |lock_|. |
| 77 double ProvideInput(media::AudioBus* audio_bus, | 77 double ProvideInput(media::AudioBus* audio_bus, |
| 78 base::TimeDelta buffer_delay) override; | 78 uint32_t frames_delayed) override; |
| 79 | 79 |
| 80 // Method to allow the unittests to inject its own sink parameters to avoid | 80 // Method to allow the unittests to inject its own sink parameters to avoid |
| 81 // query the hardware. | 81 // query the hardware. |
| 82 // TODO(xians,tommi): Remove and instead offer a way to inject the sink | 82 // TODO(xians,tommi): Remove and instead offer a way to inject the sink |
| 83 // parameters so that the implementation doesn't rely on the global default | 83 // parameters so that the implementation doesn't rely on the global default |
| 84 // hardware config but instead gets the parameters directly from the sink | 84 // hardware config but instead gets the parameters directly from the sink |
| 85 // (WebAudio in this case). Ideally the unit test should be able to use that | 85 // (WebAudio in this case). Ideally the unit test should be able to use that |
| 86 // same mechanism to inject the sink parameters for testing. | 86 // same mechanism to inject the sink parameters for testing. |
| 87 void SetSinkParamsForTesting(const media::AudioParameters& sink_params); | 87 void SetSinkParamsForTesting(const media::AudioParameters& sink_params); |
| 88 | 88 |
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| 108 | 108 |
| 109 // Flag to tell if the track has been stopped or not. | 109 // Flag to tell if the track has been stopped or not. |
| 110 bool track_stopped_; | 110 bool track_stopped_; |
| 111 | 111 |
| 112 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioSourceProvider); | 112 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioSourceProvider); |
| 113 }; | 113 }; |
| 114 | 114 |
| 115 } // namespace content | 115 } // namespace content |
| 116 | 116 |
| 117 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_SOURCE_PROVIDER_H_ | 117 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_SOURCE_PROVIDER_H_ |
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