OLD | NEW |
---|---|
1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_ | 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_ |
12 #define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_ | 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_ |
13 | 13 |
14 #include <list> | 14 #include <list> |
15 #include <memory> | 15 #include <memory> |
16 #include <string> | 16 #include <string> |
17 #include <vector> | 17 #include <vector> |
18 | 18 |
19 #include "webrtc/base/criticalsection.h" | 19 #include "webrtc/base/criticalsection.h" |
20 #include "webrtc/base/function_view.h" | 20 #include "webrtc/base/function_view.h" |
21 #include "webrtc/base/gtest_prod_util.h" | 21 #include "webrtc/base/gtest_prod_util.h" |
22 #include "webrtc/base/ignore_wundef.h" | 22 #include "webrtc/base/ignore_wundef.h" |
23 #include "webrtc/base/swap_queue.h" | 23 #include "webrtc/base/swap_queue.h" |
24 #include "webrtc/base/thread_annotations.h" | 24 #include "webrtc/base/thread_annotations.h" |
25 #include "webrtc/modules/audio_processing/aec_dumper/aec_dumper.h" | |
25 #include "webrtc/modules/audio_processing/audio_buffer.h" | 26 #include "webrtc/modules/audio_processing/audio_buffer.h" |
26 #include "webrtc/modules/audio_processing/include/audio_processing.h" | 27 #include "webrtc/modules/audio_processing/include/audio_processing.h" |
27 #include "webrtc/modules/audio_processing/render_queue_item_verifier.h" | 28 #include "webrtc/modules/audio_processing/render_queue_item_verifier.h" |
28 #include "webrtc/modules/audio_processing/rms_level.h" | 29 #include "webrtc/modules/audio_processing/rms_level.h" |
29 #include "webrtc/system_wrappers/include/file_wrapper.h" | 30 #include "webrtc/system_wrappers/include/file_wrapper.h" |
30 | 31 |
31 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP | 32 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP |
32 // Files generated at build-time by the protobuf compiler. | 33 // Files generated at build-time by the protobuf compiler. |
33 RTC_PUSH_IGNORING_WUNDEF() | 34 RTC_PUSH_IGNORING_WUNDEF() |
34 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD | 35 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD |
(...skipping 24 matching lines...) Expand all Loading... | |
59 int Initialize(int capture_input_sample_rate_hz, | 60 int Initialize(int capture_input_sample_rate_hz, |
60 int capture_output_sample_rate_hz, | 61 int capture_output_sample_rate_hz, |
61 int render_sample_rate_hz, | 62 int render_sample_rate_hz, |
62 ChannelLayout capture_input_layout, | 63 ChannelLayout capture_input_layout, |
63 ChannelLayout capture_output_layout, | 64 ChannelLayout capture_output_layout, |
64 ChannelLayout render_input_layout) override; | 65 ChannelLayout render_input_layout) override; |
65 int Initialize(const ProcessingConfig& processing_config) override; | 66 int Initialize(const ProcessingConfig& processing_config) override; |
66 void ApplyConfig(const AudioProcessing::Config& config) override; | 67 void ApplyConfig(const AudioProcessing::Config& config) override; |
67 void SetExtraOptions(const webrtc::Config& config) override; | 68 void SetExtraOptions(const webrtc::Config& config) override; |
68 void UpdateHistogramsOnCallEnd() override; | 69 void UpdateHistogramsOnCallEnd() override; |
69 int StartDebugRecording(const char filename[kMaxFilenameSize], | 70 int StartDebugRecording(const char filename[kMaxFilenameSize], |
the sun
2017/04/03 14:33:02
I think we only need one method
StartAecDump(std:
aleloi
2017/04/06 15:46:11
Good idea! I can't get rid of the other StartDebug
| |
70 int64_t max_log_size_bytes) override; | 71 int64_t max_log_size_bytes, |
71 int StartDebugRecording(FILE* handle, int64_t max_log_size_bytes) override; | 72 rtc::TaskQueue* worker_queue) override; |
72 int StartDebugRecording(FILE* handle) override; | 73 int StartDebugRecording(FILE* handle, |
73 int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override; | 74 int64_t max_log_size_bytes, |
75 rtc::TaskQueue* worker_queue) override; | |
76 int StartDebugRecording(FILE* handle, rtc::TaskQueue* worker_queue) override; | |
77 int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle, | |
78 rtc::TaskQueue* worker_queue) override; | |
74 int StopDebugRecording() override; | 79 int StopDebugRecording() override; |
75 | 80 |
76 // Capture-side exclusive methods possibly running APM in a | 81 // Capture-side exclusive methods possibly running APM in a |
77 // multi-threaded manner. Acquire the capture lock. | 82 // multi-threaded manner. Acquire the capture lock. |
78 int ProcessStream(AudioFrame* frame) override; | 83 int ProcessStream(AudioFrame* frame) override; |
79 int ProcessStream(const float* const* src, | 84 int ProcessStream(const float* const* src, |
80 size_t samples_per_channel, | 85 size_t samples_per_channel, |
81 int input_sample_rate_hz, | 86 int input_sample_rate_hz, |
82 ChannelLayout input_layout, | 87 ChannelLayout input_layout, |
83 int output_sample_rate_hz, | 88 int output_sample_rate_hz, |
(...skipping 208 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
292 int WriteConfigMessage(bool forced) EXCLUSIVE_LOCKS_REQUIRED(crit_capture_) | 297 int WriteConfigMessage(bool forced) EXCLUSIVE_LOCKS_REQUIRED(crit_capture_) |
293 EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); | 298 EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
294 | 299 |
295 // Critical section. | 300 // Critical section. |
296 rtc::CriticalSection crit_debug_; | 301 rtc::CriticalSection crit_debug_; |
297 | 302 |
298 // Debug dump state. | 303 // Debug dump state. |
299 ApmDebugDumpState debug_dump_; | 304 ApmDebugDumpState debug_dump_; |
300 #endif | 305 #endif |
301 | 306 |
307 // TODO(aleloi) doc. | |
peah-webrtc
2017/03/31 07:24:43
Please change the comment according to the style g
aleloi
2017/04/06 15:46:11
Done.
| |
308 std::unique_ptr<AecDumper> aec_dumper_; | |
peah-webrtc
2017/03/31 07:24:43
Won't this be accessed concurrently? In that case,
aleloi
2017/04/06 15:46:11
The implementation is actually thread-safe. The me
aleloi
2017/04/12 11:05:29
I've added locks for writing to aec_dump_, and an
| |
302 // Critical sections. | 309 // Critical sections. |
303 rtc::CriticalSection crit_render_ ACQUIRED_BEFORE(crit_capture_); | 310 rtc::CriticalSection crit_render_ ACQUIRED_BEFORE(crit_capture_); |
304 rtc::CriticalSection crit_capture_; | 311 rtc::CriticalSection crit_capture_; |
305 | 312 |
306 // Struct containing the Config specifying the behavior of APM. | 313 // Struct containing the Config specifying the behavior of APM. |
307 AudioProcessing::Config config_; | 314 AudioProcessing::Config config_; |
308 | 315 |
309 // Class containing information about what submodules are active. | 316 // Class containing information about what submodules are active. |
310 ApmSubmoduleStates submodule_states_; | 317 ApmSubmoduleStates submodule_states_; |
311 | 318 |
(...skipping 116 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
428 std::unique_ptr< | 435 std::unique_ptr< |
429 SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>> | 436 SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>> |
430 agc_render_signal_queue_; | 437 agc_render_signal_queue_; |
431 std::unique_ptr<SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>> | 438 std::unique_ptr<SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>> |
432 red_render_signal_queue_; | 439 red_render_signal_queue_; |
433 }; | 440 }; |
434 | 441 |
435 } // namespace webrtc | 442 } // namespace webrtc |
436 | 443 |
437 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_ | 444 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_ |
OLD | NEW |