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niklase@google.com470e71d2011-07-07 08:21:25 +00001/*
andrew@webrtc.org6f9f8172012-03-06 19:03:39 +00002 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
niklase@google.com470e71d2011-07-07 08:21:25 +00003 *
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
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
pbos@webrtc.org788acd12014-12-15 09:41:24 +000011#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
12#define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
niklase@google.com470e71d2011-07-07 08:21:25 +000013
14#include <list>
ajm@google.com808e0e02011-08-03 21:08:51 +000015#include <string>
Michael Graczyk86c6d332015-07-23 11:41:39 -070016#include <vector>
niklase@google.com470e71d2011-07-07 08:21:25 +000017
peahdf3efa82015-11-28 12:35:15 -080018#include "webrtc/base/criticalsection.h"
kwiberg@webrtc.org00b8f6b2015-02-26 14:34:55 +000019#include "webrtc/base/scoped_ptr.h"
pbos@webrtc.org788acd12014-12-15 09:41:24 +000020#include "webrtc/base/thread_annotations.h"
peahdf3efa82015-11-28 12:35:15 -080021#include "webrtc/modules/audio_processing/audio_buffer.h"
Michael Graczykdfa36052015-03-25 16:37:27 -070022#include "webrtc/modules/audio_processing/include/audio_processing.h"
peahdf3efa82015-11-28 12:35:15 -080023#include "webrtc/system_wrappers/include/file_wrapper.h"
24
25#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
26// Files generated at build-time by the protobuf compiler.
27#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
28#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
29#else
30#include "webrtc/audio_processing/debug.pb.h"
31#endif
32#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
niklase@google.com470e71d2011-07-07 08:21:25 +000033
34namespace webrtc {
andrew@webrtc.orgddbb8a22014-04-22 21:00:04 +000035
pbos@webrtc.org788acd12014-12-15 09:41:24 +000036class AgcManagerDirect;
ekmeyerson60d9b332015-08-14 10:35:55 -070037class AudioConverter;
Michael Graczykdfa36052015-03-25 16:37:27 -070038
39template<typename T>
40class Beamformer;
41
niklase@google.com470e71d2011-07-07 08:21:25 +000042class AudioProcessingImpl : public AudioProcessing {
43 public:
peahdf3efa82015-11-28 12:35:15 -080044 // Methods forcing APM to run in a single-threaded manner.
45 // Acquires both the render and capture locks.
andrew@webrtc.orge84978f2014-01-25 02:09:06 +000046 explicit AudioProcessingImpl(const Config& config);
Michael Graczykdfa36052015-03-25 16:37:27 -070047 // AudioProcessingImpl takes ownership of beamformer.
48 AudioProcessingImpl(const Config& config, Beamformer<float>* beamformer);
niklase@google.com470e71d2011-07-07 08:21:25 +000049 virtual ~AudioProcessingImpl();
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000050 int Initialize() override;
51 int Initialize(int input_sample_rate_hz,
52 int output_sample_rate_hz,
53 int reverse_sample_rate_hz,
54 ChannelLayout input_layout,
55 ChannelLayout output_layout,
56 ChannelLayout reverse_layout) override;
Michael Graczyk86c6d332015-07-23 11:41:39 -070057 int Initialize(const ProcessingConfig& processing_config) override;
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000058 void SetExtraOptions(const Config& config) override;
peahdf3efa82015-11-28 12:35:15 -080059 void UpdateHistogramsOnCallEnd() override;
60 int StartDebugRecording(const char filename[kMaxFilenameSize]) override;
61 int StartDebugRecording(FILE* handle) override;
62 int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override;
63 int StopDebugRecording() override;
64
65 // Capture-side exclusive methods possibly running APM in a
66 // multi-threaded manner. Acquire the capture lock.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000067 int ProcessStream(AudioFrame* frame) override;
68 int ProcessStream(const float* const* src,
Peter Kastingdce40cf2015-08-24 14:52:23 -070069 size_t samples_per_channel,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000070 int input_sample_rate_hz,
71 ChannelLayout input_layout,
72 int output_sample_rate_hz,
73 ChannelLayout output_layout,
74 float* const* dest) override;
Michael Graczyk86c6d332015-07-23 11:41:39 -070075 int ProcessStream(const float* const* src,
76 const StreamConfig& input_config,
77 const StreamConfig& output_config,
78 float* const* dest) override;
peahdf3efa82015-11-28 12:35:15 -080079 void set_output_will_be_muted(bool muted) override;
80 int set_stream_delay_ms(int delay) override;
81 void set_delay_offset_ms(int offset) override;
82 int delay_offset_ms() const override;
83 void set_stream_key_pressed(bool key_pressed) override;
peah66085be2015-12-16 02:02:20 -080084 int input_sample_rate_hz() const override;
peahdf3efa82015-11-28 12:35:15 -080085
86 // Render-side exclusive methods possibly running APM in a
87 // multi-threaded manner. Acquire the render lock.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000088 int AnalyzeReverseStream(AudioFrame* frame) override;
ekmeyerson60d9b332015-08-14 10:35:55 -070089 int ProcessReverseStream(AudioFrame* frame) override;
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000090 int AnalyzeReverseStream(const float* const* data,
Peter Kastingdce40cf2015-08-24 14:52:23 -070091 size_t samples_per_channel,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000092 int sample_rate_hz,
93 ChannelLayout layout) override;
ekmeyerson60d9b332015-08-14 10:35:55 -070094 int ProcessReverseStream(const float* const* src,
95 const StreamConfig& reverse_input_config,
96 const StreamConfig& reverse_output_config,
97 float* const* dest) override;
peahdf3efa82015-11-28 12:35:15 -080098
99 // Methods only accessed from APM submodules or
100 // from AudioProcessing tests in a single-threaded manner.
101 // Hence there is no need for locks in these.
102 int proc_sample_rate_hz() const override;
103 int proc_split_sample_rate_hz() const override;
104 int num_input_channels() const override;
105 int num_output_channels() const override;
106 int num_reverse_channels() const override;
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000107 int stream_delay_ms() const override;
peahdf3efa82015-11-28 12:35:15 -0800108 bool was_stream_delay_set() const override
109 EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
110
111 // Methods returning pointers to APM submodules.
112 // No locks are aquired in those, as those locks
113 // would offer no protection (the submodules are
114 // created only once in a single-treaded manner
115 // during APM creation).
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000116 EchoCancellation* echo_cancellation() const override;
117 EchoControlMobile* echo_control_mobile() const override;
118 GainControl* gain_control() const override;
119 HighPassFilter* high_pass_filter() const override;
120 LevelEstimator* level_estimator() const override;
121 NoiseSuppression* noise_suppression() const override;
122 VoiceDetection* voice_detection() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000123
andrew@webrtc.org60730cf2014-01-07 17:45:09 +0000124 protected:
andrew@webrtc.orga8b97372014-03-10 22:26:12 +0000125 // Overridden in a mock.
peahdf3efa82015-11-28 12:35:15 -0800126 virtual int InitializeLocked()
127 EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
andrew@webrtc.org60730cf2014-01-07 17:45:09 +0000128
niklase@google.com470e71d2011-07-07 08:21:25 +0000129 private:
peahdf3efa82015-11-28 12:35:15 -0800130 struct ApmPublicSubmodules;
131 struct ApmPrivateSubmodules;
132
133#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
134 // State for the debug dump.
135 struct ApmDebugDumpThreadState {
136 ApmDebugDumpThreadState() : event_msg(new audioproc::Event()) {}
137 rtc::scoped_ptr<audioproc::Event> event_msg; // Protobuf message.
138 std::string event_str; // Memory for protobuf serialization.
139
140 // Serialized string of last saved APM configuration.
141 std::string last_serialized_config;
142 };
143
144 struct ApmDebugDumpState {
145 ApmDebugDumpState() : debug_file(FileWrapper::Create()) {}
146 rtc::scoped_ptr<FileWrapper> debug_file;
147 ApmDebugDumpThreadState render;
148 ApmDebugDumpThreadState capture;
149 };
150#endif
151
152 // Method for modifying the formats struct that are called from both
153 // the render and capture threads. The check for whether modifications
154 // are needed is done while holding the render lock only, thereby avoiding
155 // that the capture thread blocks the render thread.
156 // The struct is modified in a single-threaded manner by holding both the
157 // render and capture locks.
158 int MaybeInitialize(const ProcessingConfig& config)
159 EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
160
161 int MaybeInitializeRender(const ProcessingConfig& processing_config)
162 EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
163
164 int MaybeInitializeCapture(const ProcessingConfig& processing_config)
165 EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
166
167 // Method for checking for the need of conversion. Accesses the formats
168 // structs in a read manner but the requirement for the render lock to be held
169 // was added as it currently anyway is always called in that manner.
170 bool rev_conversion_needed() const EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
171 bool render_check_rev_conversion_needed() const
172 EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
173
174 // Methods requiring APM running in a single-threaded manner.
175 // Are called with both the render and capture locks already
176 // acquired.
177 void InitializeExperimentalAgc()
178 EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
179 void InitializeTransient()
180 EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
181 void InitializeBeamformer()
182 EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
183 void InitializeIntelligibility()
184 EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
solenberg70f99032015-12-08 11:07:32 -0800185 void InitializeHighPassFilter()
186 EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
solenberg5e465c32015-12-08 13:22:33 -0800187 void InitializeNoiseSuppression()
188 EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
solenberg949028f2015-12-15 11:39:38 -0800189 void InitializeLevelEstimator()
190 EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
Michael Graczyk86c6d332015-07-23 11:41:39 -0700191 int InitializeLocked(const ProcessingConfig& config)
peahdf3efa82015-11-28 12:35:15 -0800192 EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
193
194 // Capture-side exclusive methods possibly running APM in a multi-threaded
195 // manner that are called with the render lock already acquired.
196 int ProcessStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
197 bool output_copy_needed(bool is_data_processed) const
198 EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
199 bool is_data_processed() const EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
200 bool synthesis_needed(bool is_data_processed) const
201 EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
202 bool analysis_needed(bool is_data_processed) const
203 EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
204 void MaybeUpdateHistograms() EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
205
206 // Render-side exclusive methods possibly running APM in a multi-threaded
207 // manner that are called with the render lock already acquired.
ekmeyerson60d9b332015-08-14 10:35:55 -0700208 // TODO(ekm): Remove once all clients updated to new interface.
peahdf3efa82015-11-28 12:35:15 -0800209 int AnalyzeReverseStreamLocked(const float* const* src,
210 const StreamConfig& input_config,
211 const StreamConfig& output_config)
212 EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
213 bool is_rev_processed() const EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
214 int ProcessReverseStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
andrew@webrtc.org17e40642014-03-04 20:58:13 +0000215
peahdf3efa82015-11-28 12:35:15 -0800216// Debug dump methods that are internal and called without locks.
217// TODO(peah): Make thread safe.
andrew@webrtc.org7bf26462011-12-03 00:03:31 +0000218#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
219 // TODO(andrew): make this more graceful. Ideally we would split this stuff
220 // out into a separate class with an "enabled" and "disabled" implementation.
peahdf3efa82015-11-28 12:35:15 -0800221 static int WriteMessageToDebugFile(FileWrapper* debug_file,
222 rtc::CriticalSection* crit_debug,
223 ApmDebugDumpThreadState* debug_state);
224 int WriteInitMessage() EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
Minyue13b96ba2015-10-03 00:39:14 +0200225
226 // Writes Config message. If not |forced|, only writes the current config if
227 // it is different from the last saved one; if |forced|, writes the config
228 // regardless of the last saved.
peahdf3efa82015-11-28 12:35:15 -0800229 int WriteConfigMessage(bool forced) EXCLUSIVE_LOCKS_REQUIRED(crit_capture_)
230 EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
Minyue13b96ba2015-10-03 00:39:14 +0200231
peahdf3efa82015-11-28 12:35:15 -0800232 // Critical section.
233 mutable rtc::CriticalSection crit_debug_;
Minyue13b96ba2015-10-03 00:39:14 +0200234
peahdf3efa82015-11-28 12:35:15 -0800235 // Debug dump state.
236 ApmDebugDumpState debug_dump_;
andrew@webrtc.org7bf26462011-12-03 00:03:31 +0000237#endif
niklase@google.com470e71d2011-07-07 08:21:25 +0000238
peahdf3efa82015-11-28 12:35:15 -0800239 // Critical sections.
240 mutable rtc::CriticalSection crit_render_ ACQUIRED_BEFORE(crit_capture_);
241 mutable rtc::CriticalSection crit_capture_;
242
243 // Structs containing the pointers to the submodules.
244 rtc::scoped_ptr<ApmPublicSubmodules> public_submodules_;
245 rtc::scoped_ptr<ApmPrivateSubmodules> private_submodules_
246 GUARDED_BY(crit_capture_);
247
peah192164e2015-11-17 02:16:45 -0800248 // State that is written to while holding both the render and capture locks
peahdf3efa82015-11-28 12:35:15 -0800249 // but can be read without any lock being held.
250 // As this is only accessed internally of APM, and all internal methods in APM
251 // either are holding the render or capture locks, this construct is safe as
252 // it is not possible to read the variables while writing them.
253 struct ApmFormatState {
254 ApmFormatState()
peah192164e2015-11-17 02:16:45 -0800255 : // Format of processing streams at input/output call sites.
peahdf3efa82015-11-28 12:35:15 -0800256 api_format({{{kSampleRate16kHz, 1, false},
257 {kSampleRate16kHz, 1, false},
258 {kSampleRate16kHz, 1, false},
259 {kSampleRate16kHz, 1, false}}}),
260 rev_proc_format(kSampleRate16kHz, 1) {}
261 ProcessingConfig api_format;
262 StreamConfig rev_proc_format;
263 } formats_;
Michael Graczyk86c6d332015-07-23 11:41:39 -0700264
peahdf3efa82015-11-28 12:35:15 -0800265 // APM constants.
266 const struct ApmConstants {
267 ApmConstants(int agc_startup_min_volume,
268 const std::vector<Point> array_geometry,
269 SphericalPointf target_direction,
270 bool use_new_agc,
271 bool intelligibility_enabled,
272 bool beamformer_enabled)
273 : // Format of processing streams at input/output call sites.
274 agc_startup_min_volume(agc_startup_min_volume),
275 array_geometry(array_geometry),
276 target_direction(target_direction),
277 use_new_agc(use_new_agc),
278 intelligibility_enabled(intelligibility_enabled),
279 beamformer_enabled(beamformer_enabled) {}
280 int agc_startup_min_volume;
281 std::vector<Point> array_geometry;
282 SphericalPointf target_direction;
283 bool use_new_agc;
284 bool intelligibility_enabled;
285 bool beamformer_enabled;
286 } constants_;
andrew@webrtc.orgddbb8a22014-04-22 21:00:04 +0000287
peahdf3efa82015-11-28 12:35:15 -0800288 struct ApmCaptureState {
289 ApmCaptureState(bool transient_suppressor_enabled)
290 : aec_system_delay_jumps(-1),
291 delay_offset_ms(0),
292 was_stream_delay_set(false),
293 last_stream_delay_ms(0),
294 last_aec_system_delay_ms(0),
295 stream_delay_jumps(-1),
296 output_will_be_muted(false),
297 key_pressed(false),
298 transient_suppressor_enabled(transient_suppressor_enabled),
299 fwd_proc_format(kSampleRate16kHz),
300 split_rate(kSampleRate16kHz) {}
301 int aec_system_delay_jumps;
302 int delay_offset_ms;
303 bool was_stream_delay_set;
304 int last_stream_delay_ms;
305 int last_aec_system_delay_ms;
306 int stream_delay_jumps;
307 bool output_will_be_muted;
308 bool key_pressed;
309 bool transient_suppressor_enabled;
310 rtc::scoped_ptr<AudioBuffer> capture_audio;
311 // Only the rate and samples fields of fwd_proc_format_ are used because the
312 // forward processing number of channels is mutable and is tracked by the
313 // capture_audio_.
314 StreamConfig fwd_proc_format;
315 int split_rate;
316 } capture_ GUARDED_BY(crit_capture_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000317
peahdf3efa82015-11-28 12:35:15 -0800318 struct ApmCaptureNonLockedState {
319 ApmCaptureNonLockedState()
320 : fwd_proc_format(kSampleRate16kHz),
321 split_rate(kSampleRate16kHz),
322 stream_delay_ms(0) {}
323 // Only the rate and samples fields of fwd_proc_format_ are used because the
324 // forward processing number of channels is mutable and is tracked by the
325 // capture_audio_.
326 StreamConfig fwd_proc_format;
327 int split_rate;
328 int stream_delay_ms;
329 } capture_nonlocked_;
andrew@webrtc.org75dd2882014-02-11 20:52:30 +0000330
peahdf3efa82015-11-28 12:35:15 -0800331 struct ApmRenderState {
332 rtc::scoped_ptr<AudioConverter> render_converter;
333 rtc::scoped_ptr<AudioBuffer> render_audio;
334 } render_ GUARDED_BY(crit_render_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000335};
andrew@webrtc.orgddbb8a22014-04-22 21:00:04 +0000336
niklase@google.com470e71d2011-07-07 08:21:25 +0000337} // namespace webrtc
338
pbos@webrtc.org788acd12014-12-15 09:41:24 +0000339#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_