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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_AUDIO_DEVICE_IMPL_H_ | 5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ |
6 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ | 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ |
7 | 7 |
8 #include <stdint.h> | 8 #include <stdint.h> |
9 | 9 |
| 10 #include <list> |
10 #include <string> | 11 #include <string> |
11 #include <vector> | 12 #include <vector> |
12 | 13 |
13 #include "base/compiler_specific.h" | 14 #include "base/compiler_specific.h" |
14 #include "base/files/file.h" | 15 #include "base/files/file.h" |
15 #include "base/logging.h" | 16 #include "base/logging.h" |
16 #include "base/macros.h" | 17 #include "base/macros.h" |
17 #include "base/memory/ref_counted.h" | 18 #include "base/memory/ref_counted.h" |
18 #include "base/memory/scoped_ptr.h" | 19 #include "base/memory/scoped_ptr.h" |
19 #include "base/threading/thread_checker.h" | 20 #include "base/threading/thread_checker.h" |
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299 int32_t PlayoutDelay(uint16_t* delay_ms) const override; | 300 int32_t PlayoutDelay(uint16_t* delay_ms) const override; |
300 int32_t RecordingDelay(uint16_t* delay_ms) const override; | 301 int32_t RecordingDelay(uint16_t* delay_ms) const override; |
301 int32_t RecordingSampleRate(uint32_t* sample_rate) const override; | 302 int32_t RecordingSampleRate(uint32_t* sample_rate) const override; |
302 int32_t PlayoutSampleRate(uint32_t* sample_rate) const override; | 303 int32_t PlayoutSampleRate(uint32_t* sample_rate) const override; |
303 | 304 |
304 public: | 305 public: |
305 // Sets the |renderer_|, returns false if |renderer_| already exists. | 306 // Sets the |renderer_|, returns false if |renderer_| already exists. |
306 // Called on the main renderer thread. | 307 // Called on the main renderer thread. |
307 bool SetAudioRenderer(WebRtcAudioRenderer* renderer); | 308 bool SetAudioRenderer(WebRtcAudioRenderer* renderer); |
308 | 309 |
309 // Adds/Removes the capturer to the ADM. | 310 // Adds/Removes the |capturer| to the ADM. Does NOT take ownership. |
| 311 // Capturers must remain valid until RemoveAudioCapturer() is called. |
310 // TODO(xians): Remove these two methods once the ADM does not need to pass | 312 // TODO(xians): Remove these two methods once the ADM does not need to pass |
311 // hardware information up to WebRtc. | 313 // hardware information up to WebRtc. |
312 void AddAudioCapturer(const scoped_refptr<WebRtcAudioCapturer>& capturer); | 314 void AddAudioCapturer(WebRtcAudioCapturer* capturer); |
313 void RemoveAudioCapturer(const scoped_refptr<WebRtcAudioCapturer>& capturer); | 315 void RemoveAudioCapturer(WebRtcAudioCapturer* capturer); |
314 | |
315 // Gets the default capturer, which is the last capturer in |capturers_|. | |
316 // The method can be called by both Libjingle thread and main render thread. | |
317 scoped_refptr<WebRtcAudioCapturer> GetDefaultCapturer() const; | |
318 | 316 |
319 // Gets paired device information of the capture device for the audio | 317 // Gets paired device information of the capture device for the audio |
320 // renderer. This is used to pass on a session id, sample rate and buffer | 318 // renderer. This is used to pass on a session id, sample rate and buffer |
321 // size to a webrtc audio renderer (either local or remote), so that audio | 319 // size to a webrtc audio renderer (either local or remote), so that audio |
322 // will be rendered to a matching output device. | 320 // will be rendered to a matching output device. |
323 // Returns true if the capture device has a paired output device, otherwise | 321 // Returns true if the capture device has a paired output device, otherwise |
324 // false. Note that if there are more than one open capture device the | 322 // false. Note that if there are more than one open capture device the |
325 // function will not be able to pick an appropriate device and return false. | 323 // function will not be able to pick an appropriate device and return false. |
326 bool GetAuthorizedDeviceInfoForAudioRenderer( | 324 bool GetAuthorizedDeviceInfoForAudioRenderer( |
327 int* session_id, int* output_sample_rate, int* output_buffer_size); | 325 int* session_id, int* output_sample_rate, int* output_buffer_size); |
328 | 326 |
329 const scoped_refptr<WebRtcAudioRenderer>& renderer() const { | 327 const scoped_refptr<WebRtcAudioRenderer>& renderer() const { |
330 return renderer_; | 328 return renderer_; |
331 } | 329 } |
332 | 330 |
333 private: | 331 private: |
334 typedef std::list<scoped_refptr<WebRtcAudioCapturer> > CapturerList; | 332 typedef std::list<WebRtcAudioCapturer*> CapturerList; |
335 typedef std::list<WebRtcPlayoutDataSource::Sink*> PlayoutDataSinkList; | 333 typedef std::list<WebRtcPlayoutDataSource::Sink*> PlayoutDataSinkList; |
336 class RenderBuffer; | 334 class RenderBuffer; |
337 | 335 |
338 // Make destructor private to ensure that we can only be deleted by Release(). | 336 // Make destructor private to ensure that we can only be deleted by Release(). |
339 ~WebRtcAudioDeviceImpl() override; | 337 ~WebRtcAudioDeviceImpl() override; |
340 | 338 |
341 // WebRtcAudioRendererSource implementation. | 339 // WebRtcAudioRendererSource implementation. |
342 | 340 |
343 // Called on the AudioOutputDevice worker thread. | 341 // Called on the AudioOutputDevice worker thread. |
344 void RenderData(media::AudioBus* audio_bus, | 342 void RenderData(media::AudioBus* audio_bus, |
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357 // Used to check methods that run on the main render thread. | 355 // Used to check methods that run on the main render thread. |
358 base::ThreadChecker main_thread_checker_; | 356 base::ThreadChecker main_thread_checker_; |
359 // Used to check methods that are called on libjingle's signaling thread. | 357 // Used to check methods that are called on libjingle's signaling thread. |
360 base::ThreadChecker signaling_thread_checker_; | 358 base::ThreadChecker signaling_thread_checker_; |
361 base::ThreadChecker worker_thread_checker_; | 359 base::ThreadChecker worker_thread_checker_; |
362 base::ThreadChecker audio_renderer_thread_checker_; | 360 base::ThreadChecker audio_renderer_thread_checker_; |
363 | 361 |
364 mutable int ref_count_; | 362 mutable int ref_count_; |
365 | 363 |
366 // List of captures which provides access to the native audio input layer | 364 // List of captures which provides access to the native audio input layer |
367 // in the browser process. | 365 // in the browser process. The last capturer in this list is considered the |
| 366 // "default capturer" by the methods implementing the |
| 367 // webrtc::AudioDeviceModule interface. |
368 CapturerList capturers_; | 368 CapturerList capturers_; |
369 | 369 |
370 // Provides access to the audio renderer in the browser process. | 370 // Provides access to the audio renderer in the browser process. |
371 scoped_refptr<WebRtcAudioRenderer> renderer_; | 371 scoped_refptr<WebRtcAudioRenderer> renderer_; |
372 | 372 |
373 // A list of raw pointer of WebRtcPlayoutDataSource::Sink objects which want | 373 // A list of raw pointer of WebRtcPlayoutDataSource::Sink objects which want |
374 // to get the playout data, the sink need to call RemovePlayoutSink() | 374 // to get the playout data, the sink need to call RemovePlayoutSink() |
375 // before it goes away. | 375 // before it goes away. |
376 PlayoutDataSinkList playout_sinks_; | 376 PlayoutDataSinkList playout_sinks_; |
377 | 377 |
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405 // Buffer used for temporary storage during render callback. | 405 // Buffer used for temporary storage during render callback. |
406 // It is only accessed by the audio render thread. | 406 // It is only accessed by the audio render thread. |
407 std::vector<int16_t> render_buffer_; | 407 std::vector<int16_t> render_buffer_; |
408 | 408 |
409 DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl); | 409 DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl); |
410 }; | 410 }; |
411 | 411 |
412 } // namespace content | 412 } // namespace content |
413 | 413 |
414 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ | 414 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ |
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