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Side by Side Diff: webrtc/modules/audio_device/android/opensles_recorder.cc

Issue 2119633004: Adds support for OpenSL ES based audio capture on Android (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixing presubmit warnings Created 4 years, 3 months ago
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1 /*
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
3 *
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
11 #include "webrtc/modules/audio_device/android/opensles_recorder.h"
12
13 #include <android/log.h>
14
15 #include "webrtc/base/arraysize.h"
16 #include "webrtc/base/checks.h"
17 #include "webrtc/base/format_macros.h"
18 #include "webrtc/base/timeutils.h"
19 #include "webrtc/modules/audio_device/android/audio_common.h"
20 #include "webrtc/modules/audio_device/android/audio_manager.h"
21 #include "webrtc/modules/audio_device/fine_audio_buffer.h"
22
23 #define TAG "OpenSLESRecorder"
24 #define ALOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, TAG, __VA_ARGS__)
25 #define ALOGD(...) __android_log_print(ANDROID_LOG_DEBUG, TAG, __VA_ARGS__)
26 #define ALOGE(...) __android_log_print(ANDROID_LOG_ERROR, TAG, __VA_ARGS__)
27 #define ALOGW(...) __android_log_print(ANDROID_LOG_WARN, TAG, __VA_ARGS__)
28 #define ALOGI(...) __android_log_print(ANDROID_LOG_INFO, TAG, __VA_ARGS__)
29
30 #define LOG_ON_ERROR(op) \
31 [](SLresult err) { \
32 if (err != SL_RESULT_SUCCESS) { \
33 ALOGE("%s:%d %s failed: %s", __FILE__, __LINE__, #op, \
34 GetSLErrorString(err)); \
35 return true; \
36 } \
37 return false; \
38 }(op)
39
40 namespace webrtc {
41
42 OpenSLESRecorder::OpenSLESRecorder(AudioManager* audio_manager)
43 : audio_manager_(audio_manager),
44 audio_parameters_(audio_manager->GetRecordAudioParameters()),
45 audio_device_buffer_(nullptr),
46 initialized_(false),
47 recording_(false),
48 engine_(nullptr),
49 recorder_(nullptr),
50 simple_buffer_queue_(nullptr),
51 buffer_index_(0),
52 last_rec_time_(0) {
53 ALOGD("ctor%s", GetThreadInfo().c_str());
54 // Detach from this thread since we want to use the checker to verify calls
55 // from the internal audio thread.
56 thread_checker_opensles_.DetachFromThread();
57 // Use native audio output parameters provided by the audio manager and
58 // define the PCM format structure.
59 pcm_format_ = CreatePCMConfiguration(audio_parameters_.channels(),
60 audio_parameters_.sample_rate(),
61 audio_parameters_.bits_per_sample());
62 }
63
64 OpenSLESRecorder::~OpenSLESRecorder() {
65 ALOGD("dtor%s", GetThreadInfo().c_str());
66 RTC_DCHECK(thread_checker_.CalledOnValidThread());
67 Terminate();
68 DestroyAudioRecorder();
69 engine_ = nullptr;
70 RTC_DCHECK(!engine_);
71 RTC_DCHECK(!recorder_);
72 RTC_DCHECK(!simple_buffer_queue_);
73 }
74
75 int OpenSLESRecorder::Init() {
76 ALOGD("Init%s", GetThreadInfo().c_str());
77 RTC_DCHECK(thread_checker_.CalledOnValidThread());
78 return 0;
79 }
80
81 int OpenSLESRecorder::Terminate() {
82 ALOGD("Terminate%s", GetThreadInfo().c_str());
83 RTC_DCHECK(thread_checker_.CalledOnValidThread());
84 StopRecording();
85 return 0;
86 }
87
88 int OpenSLESRecorder::InitRecording() {
89 ALOGD("InitRecording%s", GetThreadInfo().c_str());
90 RTC_DCHECK(thread_checker_.CalledOnValidThread());
91 RTC_DCHECK(!initialized_);
92 RTC_DCHECK(!recording_);
93 if (!ObtainEngineInterface()) {
94 ALOGE("Failed to obtain SL Engine interface");
95 return -1;
96 }
97 CreateAudioRecorder();
98 initialized_ = true;
99 buffer_index_ = 0;
100 return 0;
101 }
102
103 int OpenSLESRecorder::StartRecording() {
104 ALOGD("StartRecording%s", GetThreadInfo().c_str());
105 RTC_DCHECK(thread_checker_.CalledOnValidThread());
106 RTC_DCHECK(initialized_);
107 RTC_DCHECK(!recording_);
108 if (fine_audio_buffer_) {
109 fine_audio_buffer_->ResetRecord();
110 }
111 // Add buffers to the queue before changing state to SL_RECORDSTATE_RECORDING
112 // to ensure that recording starts as soon as the state is modified. On some
113 // devices, SLAndroidSimpleBufferQueue::Clear() used in Stop() does not flush
114 // the buffers as intended and we therefore check the number of buffers
115 // already queued first. Enqueue() can return SL_RESULT_BUFFER_INSUFFICIENT
116 // otherwise.
117 int num_buffers_in_queue = GetBufferCount();
118 for (int i = 0; i < kNumOfOpenSLESBuffers - num_buffers_in_queue; ++i) {
119 if (!EnqueueAudioBuffer()) {
120 recording_ = false;
121 return -1;
122 }
123 }
124 num_buffers_in_queue = GetBufferCount();
125 RTC_DCHECK_EQ(num_buffers_in_queue, kNumOfOpenSLESBuffers);
126 LogBufferState();
127 // Start audio recording by changing the state to SL_RECORDSTATE_RECORDING.
128 // Given that buffers are already enqueued, recording should start at once.
129 // The macro returns -1 if recording fails to start.
130 last_rec_time_ = rtc::Time();
131 if (LOG_ON_ERROR(
132 (*recorder_)->SetRecordState(recorder_, SL_RECORDSTATE_RECORDING))) {
133 return -1;
134 }
135 recording_ = (GetRecordState() == SL_RECORDSTATE_RECORDING);
136 RTC_DCHECK(recording_);
137 return 0;
138 }
139
140 int OpenSLESRecorder::StopRecording() {
141 ALOGD("StopRecording%s", GetThreadInfo().c_str());
142 RTC_DCHECK(thread_checker_.CalledOnValidThread());
143 if (!initialized_ || !recording_) {
144 return 0;
145 }
146 // Stop recording by setting the record state to SL_RECORDSTATE_STOPPED.
147 if (LOG_ON_ERROR(
148 (*recorder_)->SetRecordState(recorder_, SL_RECORDSTATE_STOPPED))) {
149 return -1;
150 }
151 // Clear the buffer queue to get rid of old data when resuming recording.
152 if (LOG_ON_ERROR((*simple_buffer_queue_)->Clear(simple_buffer_queue_))) {
153 return -1;
154 }
155 thread_checker_opensles_.DetachFromThread();
156 initialized_ = false;
157 recording_ = false;
158 return 0;
159 }
160
161 void OpenSLESRecorder::AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) {
162 ALOGD("AttachAudioBuffer");
163 RTC_DCHECK(thread_checker_.CalledOnValidThread());
164 RTC_CHECK(audio_buffer);
165 audio_device_buffer_ = audio_buffer;
166 // Ensure that the audio device buffer is informed about the native sample
167 // rate used on the recording side.
168 const int sample_rate_hz = audio_parameters_.sample_rate();
169 ALOGD("SetRecordingSampleRate(%d)", sample_rate_hz);
170 audio_device_buffer_->SetRecordingSampleRate(sample_rate_hz);
171 // Ensure that the audio device buffer is informed about the number of
172 // channels preferred by the OS on the recording side.
173 const size_t channels = audio_parameters_.channels();
174 ALOGD("SetRecordingChannels(%" PRIuS ")", channels);
175 audio_device_buffer_->SetRecordingChannels(channels);
176 // Allocated memory for internal data buffers given existing audio parameters.
177 AllocateDataBuffers();
178 }
179
180 int OpenSLESRecorder::EnableBuiltInAEC(bool enable) {
181 ALOGD("EnableBuiltInAEC(%d)", enable);
182 RTC_DCHECK(thread_checker_.CalledOnValidThread());
183 ALOGE("Not implemented");
184 return 0;
185 }
186
187 int OpenSLESRecorder::EnableBuiltInAGC(bool enable) {
188 ALOGD("EnableBuiltInAGC(%d)", enable);
189 RTC_DCHECK(thread_checker_.CalledOnValidThread());
190 ALOGE("Not implemented");
191 return 0;
192 }
193
194 int OpenSLESRecorder::EnableBuiltInNS(bool enable) {
195 ALOGD("EnableBuiltInNS(%d)", enable);
196 RTC_DCHECK(thread_checker_.CalledOnValidThread());
197 ALOGE("Not implemented");
198 return 0;
199 }
200
201 bool OpenSLESRecorder::ObtainEngineInterface() {
202 ALOGD("ObtainEngineInterface");
203 RTC_DCHECK(thread_checker_.CalledOnValidThread());
204 if (engine_)
205 return true;
206 // Get access to (or create if not already existing) the global OpenSL Engine
207 // object.
208 SLObjectItf engine_object = audio_manager_->GetOpenSLEngine();
209 if (engine_object == nullptr) {
210 ALOGE("Failed to access the global OpenSL engine");
211 return false;
212 }
213 // Get the SL Engine Interface which is implicit.
214 if (LOG_ON_ERROR(
215 (*engine_object)
216 ->GetInterface(engine_object, SL_IID_ENGINE, &engine_))) {
217 return false;
218 }
219 return true;
220 }
221
222 bool OpenSLESRecorder::CreateAudioRecorder() {
223 ALOGD("CreateAudioRecorder");
224 RTC_DCHECK(thread_checker_.CalledOnValidThread());
225 if (recorder_object_.Get())
226 return true;
227 RTC_DCHECK(!recorder_);
228 RTC_DCHECK(!simple_buffer_queue_);
229
230 // Audio source configuration.
231 SLDataLocator_IODevice mic_locator = {SL_DATALOCATOR_IODEVICE,
232 SL_IODEVICE_AUDIOINPUT,
233 SL_DEFAULTDEVICEID_AUDIOINPUT, NULL};
234 SLDataSource audio_source = {&mic_locator, NULL};
235
236 // Audio sink configuration.
237 SLDataLocator_AndroidSimpleBufferQueue buffer_queue = {
238 SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE,
239 static_cast<SLuint32>(kNumOfOpenSLESBuffers)};
240 SLDataSink audio_sink = {&buffer_queue, &pcm_format_};
241
242 // Create the audio recorder object (requires the RECORD_AUDIO permission).
243 // Do not realize the recorder yet. Set the configuration first.
244 const SLInterfaceID interface_id[] = {SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
245 SL_IID_ANDROIDCONFIGURATION};
246 const SLboolean interface_required[] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
247 if (LOG_ON_ERROR((*engine_)->CreateAudioRecorder(
248 engine_, recorder_object_.Receive(), &audio_source, &audio_sink,
249 arraysize(interface_id), interface_id, interface_required))) {
250 return false;
251 }
252
253 // Configure the audio recorder (before it is realized).
254 SLAndroidConfigurationItf recorder_config;
255 if (LOG_ON_ERROR((recorder_object_->GetInterface(recorder_object_.Get(),
256 SL_IID_ANDROIDCONFIGURATION,
257 &recorder_config)))) {
258 return false;
259 }
260
261 // Uses the default microphone tuned for audio communication.
262 // Note that, SL_ANDROID_RECORDING_PRESET_VOICE_RECOGNITION leads to a fast
263 // track but also excludes usage of required effects like AEC, AGC and NS.
264 // SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION
265 SLint32 stream_type = SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION;
266 if (LOG_ON_ERROR(((*recorder_config)
267 ->SetConfiguration(recorder_config,
268 SL_ANDROID_KEY_RECORDING_PRESET,
269 &stream_type, sizeof(SLint32))))) {
270 return false;
271 }
272
273 // The audio recorder can now be realized (in synchronous mode).
274 if (LOG_ON_ERROR((recorder_object_->Realize(recorder_object_.Get(),
275 SL_BOOLEAN_FALSE)))) {
276 return false;
277 }
278
279 // Get the implicit recorder interface (SL_IID_RECORD).
280 if (LOG_ON_ERROR((recorder_object_->GetInterface(
281 recorder_object_.Get(), SL_IID_RECORD, &recorder_)))) {
282 return false;
283 }
284
285 // Get the simple buffer queue interface (SL_IID_ANDROIDSIMPLEBUFFERQUEUE).
286 // It was explicitly requested.
287 if (LOG_ON_ERROR((recorder_object_->GetInterface(
288 recorder_object_.Get(), SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
289 &simple_buffer_queue_)))) {
290 return false;
291 }
292
293 // Register the input callback for the simple buffer queue.
294 // This callback will be called when receiving new data from the device.
295 if (LOG_ON_ERROR(((*simple_buffer_queue_)
296 ->RegisterCallback(simple_buffer_queue_,
297 SimpleBufferQueueCallback, this)))) {
298 return false;
299 }
300 return true;
301 }
302
303 void OpenSLESRecorder::DestroyAudioRecorder() {
304 ALOGD("DestroyAudioRecorder");
305 RTC_DCHECK(thread_checker_.CalledOnValidThread());
306 if (!recorder_object_.Get())
307 return;
308 (*simple_buffer_queue_)
309 ->RegisterCallback(simple_buffer_queue_, nullptr, nullptr);
310 recorder_object_.Reset();
311 recorder_ = nullptr;
312 simple_buffer_queue_ = nullptr;
313 }
314
315 void OpenSLESRecorder::SimpleBufferQueueCallback(
316 SLAndroidSimpleBufferQueueItf buffer_queue,
317 void* context) {
318 OpenSLESRecorder* stream = static_cast<OpenSLESRecorder*>(context);
319 stream->ReadBufferQueue();
320 }
321
322 void OpenSLESRecorder::AllocateDataBuffers() {
323 ALOGD("AllocateDataBuffers");
324 RTC_DCHECK(thread_checker_.CalledOnValidThread());
325 RTC_DCHECK(!simple_buffer_queue_);
326 RTC_CHECK(audio_device_buffer_);
327 // Create a modified audio buffer class which allows us to deliver any number
328 // of samples (and not only multiple of 10ms) to match the native audio unit
329 // buffer size.
330 ALOGD("frames per native buffer: %" PRIuS,
331 audio_parameters_.frames_per_buffer());
332 ALOGD("frames per 10ms buffer: %" PRIuS,
333 audio_parameters_.frames_per_10ms_buffer());
334 ALOGD("bytes per native buffer: %" PRIuS,
335 audio_parameters_.GetBytesPerBuffer());
336 ALOGD("native sample rate: %d", audio_parameters_.sample_rate());
337 RTC_DCHECK(audio_device_buffer_);
338 fine_audio_buffer_.reset(new FineAudioBuffer(
339 audio_device_buffer_, audio_parameters_.GetBytesPerBuffer(),
340 audio_parameters_.sample_rate()));
341 // Allocate queue of audio buffers that stores recorded audio samples.
342 const int data_size_bytes = audio_parameters_.GetBytesPerBuffer();
343 audio_buffers_.reset(new std::unique_ptr<SLint8[]>[kNumOfOpenSLESBuffers]);
344 for (int i = 0; i < kNumOfOpenSLESBuffers; ++i) {
345 audio_buffers_[i].reset(new SLint8[data_size_bytes]);
346 }
347 }
348
349 void OpenSLESRecorder::ReadBufferQueue() {
350 RTC_DCHECK(thread_checker_opensles_.CalledOnValidThread());
351 SLuint32 state = GetRecordState();
352 if (state != SL_RECORDSTATE_RECORDING) {
353 ALOGW("Buffer callback in non-recording state!");
354 return;
355 }
356 // Check delta time between two successive callbacks and provide a warning
357 // if it becomes very large.
358 // TODO(henrika): using 150ms as upper limit but this value is rather random.
359 const uint32_t current_time = rtc::Time();
360 const uint32_t diff = current_time - last_rec_time_;
361 if (diff > 150) {
362 ALOGW("Bad OpenSL ES record timing, dT=%u [ms]", diff);
363 }
364 last_rec_time_ = current_time;
365 // Send recorded audio data to the WebRTC sink.
366 // TODO(henrika): fix delay estimates. It is OK to use fixed values for now
367 // since there is no support to turn off built-in EC in combination with
368 // OpenSL ES anyhow. Hence, as is, the WebRTC based AEC (which would use
369 // these estimates) will never be active.
370 const size_t size_in_bytes =
371 static_cast<size_t>(audio_parameters_.GetBytesPerBuffer());
372 const int8_t* data =
373 static_cast<const int8_t*>(audio_buffers_[buffer_index_].get());
374 fine_audio_buffer_->DeliverRecordedData(data, size_in_bytes, 25, 25);
375 // Enqueue the utilized audio buffer and use if for recording again.
376 EnqueueAudioBuffer();
377 }
378
379 bool OpenSLESRecorder::EnqueueAudioBuffer() {
380 SLresult err =
381 (*simple_buffer_queue_)
382 ->Enqueue(simple_buffer_queue_, audio_buffers_[buffer_index_].get(),
383 audio_parameters_.GetBytesPerBuffer());
384 if (SL_RESULT_SUCCESS != err) {
385 ALOGE("Enqueue failed: %s", GetSLErrorString(err));
386 return false;
387 }
388 buffer_index_ = (buffer_index_ + 1) % kNumOfOpenSLESBuffers;
389 return true;
390 }
391
392 SLuint32 OpenSLESRecorder::GetRecordState() const {
393 RTC_DCHECK(recorder_);
394 SLuint32 state;
395 SLresult err = (*recorder_)->GetRecordState(recorder_, &state);
396 if (SL_RESULT_SUCCESS != err) {
397 ALOGE("GetRecordState failed: %s", GetSLErrorString(err));
398 }
399 return state;
400 }
401
402 SLAndroidSimpleBufferQueueState OpenSLESRecorder::GetBufferQueueState() const {
403 RTC_DCHECK(simple_buffer_queue_);
404 // state.count: Number of buffers currently in the queue.
405 // state.index: Index of the currently filling buffer. This is a linear index
406 // that keeps a cumulative count of the number of buffers recorded.
407 SLAndroidSimpleBufferQueueState state;
408 SLresult err =
409 (*simple_buffer_queue_)->GetState(simple_buffer_queue_, &state);
410 if (SL_RESULT_SUCCESS != err) {
411 ALOGE("GetState failed: %s", GetSLErrorString(err));
412 }
413 return state;
414 }
415
416 void OpenSLESRecorder::LogBufferState() const {
417 SLAndroidSimpleBufferQueueState state = GetBufferQueueState();
418 ALOGD("state.count:%d state.index:%d", state.count, state.index);
419 }
420
421 SLuint32 OpenSLESRecorder::GetBufferCount() {
422 SLAndroidSimpleBufferQueueState state = GetBufferQueueState();
423 return state.count;
424 }
425
426 } // namespace webrtc
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