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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;
84
85 // Render-side exclusive methods possibly running APM in a
86 // multi-threaded manner. Acquire the render lock.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000087 int AnalyzeReverseStream(AudioFrame* frame) override;
ekmeyerson60d9b332015-08-14 10:35:55 -070088 int ProcessReverseStream(AudioFrame* frame) override;
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000089 int AnalyzeReverseStream(const float* const* data,
Peter Kastingdce40cf2015-08-24 14:52:23 -070090 size_t samples_per_channel,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000091 int sample_rate_hz,
92 ChannelLayout layout) override;
ekmeyerson60d9b332015-08-14 10:35:55 -070093 int ProcessReverseStream(const float* const* src,
94 const StreamConfig& reverse_input_config,
95 const StreamConfig& reverse_output_config,
96 float* const* dest) override;
peahdf3efa82015-11-28 12:35:15 -080097
98 // Methods only accessed from APM submodules or
99 // from AudioProcessing tests in a single-threaded manner.
100 // Hence there is no need for locks in these.
101 int proc_sample_rate_hz() const override;
102 int proc_split_sample_rate_hz() const override;
103 int num_input_channels() const override;
104 int num_output_channels() const override;
105 int num_reverse_channels() const override;
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000106 int stream_delay_ms() const override;
peahdf3efa82015-11-28 12:35:15 -0800107 bool was_stream_delay_set() const override
108 EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
109
110 // Methods returning pointers to APM submodules.
111 // No locks are aquired in those, as those locks
112 // would offer no protection (the submodules are
113 // created only once in a single-treaded manner
114 // during APM creation).
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000115 EchoCancellation* echo_cancellation() const override;
116 EchoControlMobile* echo_control_mobile() const override;
117 GainControl* gain_control() const override;
118 HighPassFilter* high_pass_filter() const override;
119 LevelEstimator* level_estimator() const override;
120 NoiseSuppression* noise_suppression() const override;
121 VoiceDetection* voice_detection() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000122
andrew@webrtc.org60730cf2014-01-07 17:45:09 +0000123 protected:
andrew@webrtc.orga8b97372014-03-10 22:26:12 +0000124 // Overridden in a mock.
peahdf3efa82015-11-28 12:35:15 -0800125 virtual int InitializeLocked()
126 EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
andrew@webrtc.org60730cf2014-01-07 17:45:09 +0000127
niklase@google.com470e71d2011-07-07 08:21:25 +0000128 private:
peahdf3efa82015-11-28 12:35:15 -0800129 struct ApmPublicSubmodules;
130 struct ApmPrivateSubmodules;
131
132#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
133 // State for the debug dump.
134 struct ApmDebugDumpThreadState {
135 ApmDebugDumpThreadState() : event_msg(new audioproc::Event()) {}
136 rtc::scoped_ptr<audioproc::Event> event_msg; // Protobuf message.
137 std::string event_str; // Memory for protobuf serialization.
138
139 // Serialized string of last saved APM configuration.
140 std::string last_serialized_config;
141 };
142
143 struct ApmDebugDumpState {
144 ApmDebugDumpState() : debug_file(FileWrapper::Create()) {}
145 rtc::scoped_ptr<FileWrapper> debug_file;
146 ApmDebugDumpThreadState render;
147 ApmDebugDumpThreadState capture;
148 };
149#endif
150
151 // Method for modifying the formats struct that are called from both
152 // the render and capture threads. The check for whether modifications
153 // are needed is done while holding the render lock only, thereby avoiding
154 // that the capture thread blocks the render thread.
155 // The struct is modified in a single-threaded manner by holding both the
156 // render and capture locks.
157 int MaybeInitialize(const ProcessingConfig& config)
158 EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
159
160 int MaybeInitializeRender(const ProcessingConfig& processing_config)
161 EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
162
163 int MaybeInitializeCapture(const ProcessingConfig& processing_config)
164 EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
165
166 // Method for checking for the need of conversion. Accesses the formats
167 // structs in a read manner but the requirement for the render lock to be held
168 // was added as it currently anyway is always called in that manner.
169 bool rev_conversion_needed() const EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
170 bool render_check_rev_conversion_needed() const
171 EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
172
173 // Methods requiring APM running in a single-threaded manner.
174 // Are called with both the render and capture locks already
175 // acquired.
176 void InitializeExperimentalAgc()
177 EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
178 void InitializeTransient()
179 EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
180 void InitializeBeamformer()
181 EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
182 void InitializeIntelligibility()
183 EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
solenberg70f99032015-12-08 11:07:32 -0800184 void InitializeHighPassFilter()
185 EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
solenberg5e465c32015-12-08 13:22:33 -0800186 void InitializeNoiseSuppression()
187 EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
solenberg949028f2015-12-15 11:39:38 -0800188 void InitializeLevelEstimator()
189 EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
Michael Graczyk86c6d332015-07-23 11:41:39 -0700190 int InitializeLocked(const ProcessingConfig& config)
peahdf3efa82015-11-28 12:35:15 -0800191 EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
192
193 // Capture-side exclusive methods possibly running APM in a multi-threaded
194 // manner that are called with the render lock already acquired.
195 int ProcessStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
196 bool output_copy_needed(bool is_data_processed) const
197 EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
198 bool is_data_processed() const EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
199 bool synthesis_needed(bool is_data_processed) const
200 EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
201 bool analysis_needed(bool is_data_processed) const
202 EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
203 void MaybeUpdateHistograms() EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
204
205 // Render-side exclusive methods possibly running APM in a multi-threaded
206 // manner that are called with the render lock already acquired.
ekmeyerson60d9b332015-08-14 10:35:55 -0700207 // TODO(ekm): Remove once all clients updated to new interface.
peahdf3efa82015-11-28 12:35:15 -0800208 int AnalyzeReverseStreamLocked(const float* const* src,
209 const StreamConfig& input_config,
210 const StreamConfig& output_config)
211 EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
212 bool is_rev_processed() const EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
213 int ProcessReverseStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
andrew@webrtc.org17e40642014-03-04 20:58:13 +0000214
peahdf3efa82015-11-28 12:35:15 -0800215// Debug dump methods that are internal and called without locks.
216// TODO(peah): Make thread safe.
andrew@webrtc.org7bf26462011-12-03 00:03:31 +0000217#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
218 // TODO(andrew): make this more graceful. Ideally we would split this stuff
219 // out into a separate class with an "enabled" and "disabled" implementation.
peahdf3efa82015-11-28 12:35:15 -0800220 static int WriteMessageToDebugFile(FileWrapper* debug_file,
221 rtc::CriticalSection* crit_debug,
222 ApmDebugDumpThreadState* debug_state);
223 int WriteInitMessage() EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
Minyue13b96ba2015-10-03 00:39:14 +0200224
225 // Writes Config message. If not |forced|, only writes the current config if
226 // it is different from the last saved one; if |forced|, writes the config
227 // regardless of the last saved.
peahdf3efa82015-11-28 12:35:15 -0800228 int WriteConfigMessage(bool forced) EXCLUSIVE_LOCKS_REQUIRED(crit_capture_)
229 EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
Minyue13b96ba2015-10-03 00:39:14 +0200230
peahdf3efa82015-11-28 12:35:15 -0800231 // Critical section.
232 mutable rtc::CriticalSection crit_debug_;
Minyue13b96ba2015-10-03 00:39:14 +0200233
peahdf3efa82015-11-28 12:35:15 -0800234 // Debug dump state.
235 ApmDebugDumpState debug_dump_;
andrew@webrtc.org7bf26462011-12-03 00:03:31 +0000236#endif
niklase@google.com470e71d2011-07-07 08:21:25 +0000237
peahdf3efa82015-11-28 12:35:15 -0800238 // Critical sections.
239 mutable rtc::CriticalSection crit_render_ ACQUIRED_BEFORE(crit_capture_);
240 mutable rtc::CriticalSection crit_capture_;
241
242 // Structs containing the pointers to the submodules.
243 rtc::scoped_ptr<ApmPublicSubmodules> public_submodules_;
244 rtc::scoped_ptr<ApmPrivateSubmodules> private_submodules_
245 GUARDED_BY(crit_capture_);
246
peah192164e2015-11-17 02:16:45 -0800247 // State that is written to while holding both the render and capture locks
peahdf3efa82015-11-28 12:35:15 -0800248 // but can be read without any lock being held.
249 // As this is only accessed internally of APM, and all internal methods in APM
250 // either are holding the render or capture locks, this construct is safe as
251 // it is not possible to read the variables while writing them.
252 struct ApmFormatState {
253 ApmFormatState()
peah192164e2015-11-17 02:16:45 -0800254 : // Format of processing streams at input/output call sites.
peahdf3efa82015-11-28 12:35:15 -0800255 api_format({{{kSampleRate16kHz, 1, false},
256 {kSampleRate16kHz, 1, false},
257 {kSampleRate16kHz, 1, false},
258 {kSampleRate16kHz, 1, false}}}),
259 rev_proc_format(kSampleRate16kHz, 1) {}
260 ProcessingConfig api_format;
261 StreamConfig rev_proc_format;
262 } formats_;
Michael Graczyk86c6d332015-07-23 11:41:39 -0700263
peahdf3efa82015-11-28 12:35:15 -0800264 // APM constants.
265 const struct ApmConstants {
266 ApmConstants(int agc_startup_min_volume,
267 const std::vector<Point> array_geometry,
268 SphericalPointf target_direction,
269 bool use_new_agc,
270 bool intelligibility_enabled,
271 bool beamformer_enabled)
272 : // Format of processing streams at input/output call sites.
273 agc_startup_min_volume(agc_startup_min_volume),
274 array_geometry(array_geometry),
275 target_direction(target_direction),
276 use_new_agc(use_new_agc),
277 intelligibility_enabled(intelligibility_enabled),
278 beamformer_enabled(beamformer_enabled) {}
279 int agc_startup_min_volume;
280 std::vector<Point> array_geometry;
281 SphericalPointf target_direction;
282 bool use_new_agc;
283 bool intelligibility_enabled;
284 bool beamformer_enabled;
285 } constants_;
andrew@webrtc.orgddbb8a22014-04-22 21:00:04 +0000286
peahdf3efa82015-11-28 12:35:15 -0800287 struct ApmCaptureState {
288 ApmCaptureState(bool transient_suppressor_enabled)
289 : aec_system_delay_jumps(-1),
290 delay_offset_ms(0),
291 was_stream_delay_set(false),
292 last_stream_delay_ms(0),
293 last_aec_system_delay_ms(0),
294 stream_delay_jumps(-1),
295 output_will_be_muted(false),
296 key_pressed(false),
297 transient_suppressor_enabled(transient_suppressor_enabled),
298 fwd_proc_format(kSampleRate16kHz),
299 split_rate(kSampleRate16kHz) {}
300 int aec_system_delay_jumps;
301 int delay_offset_ms;
302 bool was_stream_delay_set;
303 int last_stream_delay_ms;
304 int last_aec_system_delay_ms;
305 int stream_delay_jumps;
306 bool output_will_be_muted;
307 bool key_pressed;
308 bool transient_suppressor_enabled;
309 rtc::scoped_ptr<AudioBuffer> capture_audio;
310 // Only the rate and samples fields of fwd_proc_format_ are used because the
311 // forward processing number of channels is mutable and is tracked by the
312 // capture_audio_.
313 StreamConfig fwd_proc_format;
314 int split_rate;
315 } capture_ GUARDED_BY(crit_capture_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000316
peahdf3efa82015-11-28 12:35:15 -0800317 struct ApmCaptureNonLockedState {
318 ApmCaptureNonLockedState()
319 : fwd_proc_format(kSampleRate16kHz),
320 split_rate(kSampleRate16kHz),
321 stream_delay_ms(0) {}
322 // Only the rate and samples fields of fwd_proc_format_ are used because the
323 // forward processing number of channels is mutable and is tracked by the
324 // capture_audio_.
325 StreamConfig fwd_proc_format;
326 int split_rate;
327 int stream_delay_ms;
328 } capture_nonlocked_;
andrew@webrtc.org75dd2882014-02-11 20:52:30 +0000329
peahdf3efa82015-11-28 12:35:15 -0800330 struct ApmRenderState {
331 rtc::scoped_ptr<AudioConverter> render_converter;
332 rtc::scoped_ptr<AudioBuffer> render_audio;
333 } render_ GUARDED_BY(crit_render_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000334};
andrew@webrtc.orgddbb8a22014-04-22 21:00:04 +0000335
niklase@google.com470e71d2011-07-07 08:21:25 +0000336} // namespace webrtc
337
pbos@webrtc.org788acd12014-12-15 09:41:24 +0000338#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_