1/* 2 * Copyright (C) 2010 The Android Open Source Project 3 * 4 * Licensed under the Apache License, Version 2.0 (the "License"); 5 * you may not use this file except in compliance with the License. 6 * You may obtain a copy of the License at 7 * 8 * http://www.apache.org/licenses/LICENSE-2.0 9 * 10 * Unless required by applicable law or agreed to in writing, software 11 * distributed under the License is distributed on an "AS IS" BASIS, 12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 13 * See the License for the specific language governing permissions and 14 * limitations under the License. 15 */ 16 17#include <inttypes.h> 18#include <stdlib.h> 19 20//#define LOG_NDEBUG 0 21#define LOG_TAG "AudioSource" 22#include <utils/Log.h> 23 24#include <media/AudioRecord.h> 25#include <media/stagefright/AudioSource.h> 26#include <media/stagefright/MediaBuffer.h> 27#include <media/stagefright/MediaDefs.h> 28#include <media/stagefright/MetaData.h> 29#include <media/stagefright/foundation/ADebug.h> 30#include <media/stagefright/foundation/ALooper.h> 31#include <cutils/properties.h> 32 33namespace android { 34 35static void AudioRecordCallbackFunction(int event, void *user, void *info) { 36 AudioSource *source = (AudioSource *) user; 37 switch (event) { 38 case AudioRecord::EVENT_MORE_DATA: { 39 source->dataCallback(*((AudioRecord::Buffer *) info)); 40 break; 41 } 42 case AudioRecord::EVENT_OVERRUN: { 43 ALOGW("AudioRecord reported overrun!"); 44 break; 45 } 46 default: 47 // does nothing 48 break; 49 } 50} 51 52AudioSource::AudioSource( 53 audio_source_t inputSource, uint32_t sampleRate, uint32_t channelCount) 54 : mStarted(false), 55 mSampleRate(sampleRate), 56 mPrevSampleTimeUs(0), 57 mNumFramesReceived(0), 58 mNumClientOwnedBuffers(0) { 59 ALOGV("sampleRate: %d, channelCount: %d", sampleRate, channelCount); 60 CHECK(channelCount == 1 || channelCount == 2); 61 62 size_t minFrameCount; 63 status_t status = AudioRecord::getMinFrameCount(&minFrameCount, 64 sampleRate, 65 AUDIO_FORMAT_PCM_16_BIT, 66 audio_channel_in_mask_from_count(channelCount)); 67 if (status == OK) { 68 // make sure that the AudioRecord callback never returns more than the maximum 69 // buffer size 70 uint32_t frameCount = kMaxBufferSize / sizeof(int16_t) / channelCount; 71 72 // make sure that the AudioRecord total buffer size is large enough 73 size_t bufCount = 2; 74 while ((bufCount * frameCount) < minFrameCount) { 75 bufCount++; 76 } 77 78 mRecord = new AudioRecord( 79 inputSource, sampleRate, AUDIO_FORMAT_PCM_16_BIT, 80 audio_channel_in_mask_from_count(channelCount), 81 (size_t) (bufCount * frameCount), 82 AudioRecordCallbackFunction, 83 this, 84 frameCount /*notificationFrames*/); 85 mInitCheck = mRecord->initCheck(); 86 } else { 87 mInitCheck = status; 88 } 89} 90 91AudioSource::~AudioSource() { 92 if (mStarted) { 93 reset(); 94 } 95} 96 97status_t AudioSource::initCheck() const { 98 return mInitCheck; 99} 100 101status_t AudioSource::start(MetaData *params) { 102 Mutex::Autolock autoLock(mLock); 103 if (mStarted) { 104 return UNKNOWN_ERROR; 105 } 106 107 if (mInitCheck != OK) { 108 return NO_INIT; 109 } 110 111 mTrackMaxAmplitude = false; 112 mMaxAmplitude = 0; 113 mInitialReadTimeUs = 0; 114 mStartTimeUs = 0; 115 int64_t startTimeUs; 116 if (params && params->findInt64(kKeyTime, &startTimeUs)) { 117 mStartTimeUs = startTimeUs; 118 } 119 status_t err = mRecord->start(); 120 if (err == OK) { 121 mStarted = true; 122 } else { 123 mRecord.clear(); 124 } 125 126 127 return err; 128} 129 130void AudioSource::releaseQueuedFrames_l() { 131 ALOGV("releaseQueuedFrames_l"); 132 List<MediaBuffer *>::iterator it; 133 while (!mBuffersReceived.empty()) { 134 it = mBuffersReceived.begin(); 135 (*it)->release(); 136 mBuffersReceived.erase(it); 137 } 138} 139 140void AudioSource::waitOutstandingEncodingFrames_l() { 141 ALOGV("waitOutstandingEncodingFrames_l: %" PRId64, mNumClientOwnedBuffers); 142 while (mNumClientOwnedBuffers > 0) { 143 mFrameEncodingCompletionCondition.wait(mLock); 144 } 145} 146 147status_t AudioSource::reset() { 148 Mutex::Autolock autoLock(mLock); 149 if (!mStarted) { 150 return UNKNOWN_ERROR; 151 } 152 153 if (mInitCheck != OK) { 154 return NO_INIT; 155 } 156 157 mStarted = false; 158 mFrameAvailableCondition.signal(); 159 160 mRecord->stop(); 161 waitOutstandingEncodingFrames_l(); 162 releaseQueuedFrames_l(); 163 164 return OK; 165} 166 167sp<MetaData> AudioSource::getFormat() { 168 Mutex::Autolock autoLock(mLock); 169 if (mInitCheck != OK) { 170 return 0; 171 } 172 173 sp<MetaData> meta = new MetaData; 174 meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_RAW); 175 meta->setInt32(kKeySampleRate, mSampleRate); 176 meta->setInt32(kKeyChannelCount, mRecord->channelCount()); 177 meta->setInt32(kKeyMaxInputSize, kMaxBufferSize); 178 179 return meta; 180} 181 182void AudioSource::rampVolume( 183 int32_t startFrame, int32_t rampDurationFrames, 184 uint8_t *data, size_t bytes) { 185 186 const int32_t kShift = 14; 187 int32_t fixedMultiplier = (startFrame << kShift) / rampDurationFrames; 188 const int32_t nChannels = mRecord->channelCount(); 189 int32_t stopFrame = startFrame + bytes / sizeof(int16_t); 190 int16_t *frame = (int16_t *) data; 191 if (stopFrame > rampDurationFrames) { 192 stopFrame = rampDurationFrames; 193 } 194 195 while (startFrame < stopFrame) { 196 if (nChannels == 1) { // mono 197 frame[0] = (frame[0] * fixedMultiplier) >> kShift; 198 ++frame; 199 ++startFrame; 200 } else { // stereo 201 frame[0] = (frame[0] * fixedMultiplier) >> kShift; 202 frame[1] = (frame[1] * fixedMultiplier) >> kShift; 203 frame += 2; 204 startFrame += 2; 205 } 206 207 // Update the multiplier every 4 frames 208 if ((startFrame & 3) == 0) { 209 fixedMultiplier = (startFrame << kShift) / rampDurationFrames; 210 } 211 } 212} 213 214status_t AudioSource::read( 215 MediaBuffer **out, const ReadOptions * /* options */) { 216 Mutex::Autolock autoLock(mLock); 217 *out = NULL; 218 219 if (mInitCheck != OK) { 220 return NO_INIT; 221 } 222 223 while (mStarted && mBuffersReceived.empty()) { 224 mFrameAvailableCondition.wait(mLock); 225 } 226 if (!mStarted) { 227 return OK; 228 } 229 MediaBuffer *buffer = *mBuffersReceived.begin(); 230 mBuffersReceived.erase(mBuffersReceived.begin()); 231 ++mNumClientOwnedBuffers; 232 buffer->setObserver(this); 233 buffer->add_ref(); 234 235 // Mute/suppress the recording sound 236 int64_t timeUs; 237 CHECK(buffer->meta_data()->findInt64(kKeyTime, &timeUs)); 238 int64_t elapsedTimeUs = timeUs - mStartTimeUs; 239 if (elapsedTimeUs < kAutoRampStartUs) { 240 memset((uint8_t *) buffer->data(), 0, buffer->range_length()); 241 } else if (elapsedTimeUs < kAutoRampStartUs + kAutoRampDurationUs) { 242 int32_t autoRampDurationFrames = 243 ((int64_t)kAutoRampDurationUs * mSampleRate + 500000LL) / 1000000LL; //Need type casting 244 245 int32_t autoRampStartFrames = 246 ((int64_t)kAutoRampStartUs * mSampleRate + 500000LL) / 1000000LL; //Need type casting 247 248 int32_t nFrames = mNumFramesReceived - autoRampStartFrames; 249 rampVolume(nFrames, autoRampDurationFrames, 250 (uint8_t *) buffer->data(), buffer->range_length()); 251 } 252 253 // Track the max recording signal amplitude. 254 if (mTrackMaxAmplitude) { 255 trackMaxAmplitude( 256 (int16_t *) buffer->data(), buffer->range_length() >> 1); 257 } 258 259 *out = buffer; 260 return OK; 261} 262 263void AudioSource::signalBufferReturned(MediaBuffer *buffer) { 264 ALOGV("signalBufferReturned: %p", buffer->data()); 265 Mutex::Autolock autoLock(mLock); 266 --mNumClientOwnedBuffers; 267 buffer->setObserver(0); 268 buffer->release(); 269 mFrameEncodingCompletionCondition.signal(); 270 return; 271} 272 273status_t AudioSource::dataCallback(const AudioRecord::Buffer& audioBuffer) { 274 int64_t timeUs = systemTime() / 1000ll; 275 276 ALOGV("dataCallbackTimestamp: %" PRId64 " us", timeUs); 277 Mutex::Autolock autoLock(mLock); 278 if (!mStarted) { 279 ALOGW("Spurious callback from AudioRecord. Drop the audio data."); 280 return OK; 281 } 282 283 // Drop retrieved and previously lost audio data. 284 if (mNumFramesReceived == 0 && timeUs < mStartTimeUs) { 285 (void) mRecord->getInputFramesLost(); 286 ALOGV("Drop audio data at %" PRId64 "/%" PRId64 " us", timeUs, mStartTimeUs); 287 return OK; 288 } 289 290 if (mNumFramesReceived == 0 && mPrevSampleTimeUs == 0) { 291 mInitialReadTimeUs = timeUs; 292 // Initial delay 293 if (mStartTimeUs > 0) { 294 mStartTimeUs = timeUs - mStartTimeUs; 295 } else { 296 // Assume latency is constant. 297 mStartTimeUs += mRecord->latency() * 1000; 298 } 299 300 mPrevSampleTimeUs = mStartTimeUs; 301 } 302 303 size_t numLostBytes = 0; 304 if (mNumFramesReceived > 0) { // Ignore earlier frame lost 305 // getInputFramesLost() returns the number of lost frames. 306 // Convert number of frames lost to number of bytes lost. 307 numLostBytes = mRecord->getInputFramesLost() * mRecord->frameSize(); 308 } 309 310 CHECK_EQ(numLostBytes & 1, 0u); 311 CHECK_EQ(audioBuffer.size & 1, 0u); 312 if (numLostBytes > 0) { 313 // Loss of audio frames should happen rarely; thus the LOGW should 314 // not cause a logging spam 315 ALOGW("Lost audio record data: %zu bytes", numLostBytes); 316 } 317 318 while (numLostBytes > 0) { 319 size_t bufferSize = numLostBytes; 320 if (numLostBytes > kMaxBufferSize) { 321 numLostBytes -= kMaxBufferSize; 322 bufferSize = kMaxBufferSize; 323 } else { 324 numLostBytes = 0; 325 } 326 MediaBuffer *lostAudioBuffer = new MediaBuffer(bufferSize); 327 memset(lostAudioBuffer->data(), 0, bufferSize); 328 lostAudioBuffer->set_range(0, bufferSize); 329 queueInputBuffer_l(lostAudioBuffer, timeUs); 330 } 331 332 if (audioBuffer.size == 0) { 333 ALOGW("Nothing is available from AudioRecord callback buffer"); 334 return OK; 335 } 336 337 const size_t bufferSize = audioBuffer.size; 338 MediaBuffer *buffer = new MediaBuffer(bufferSize); 339 memcpy((uint8_t *) buffer->data(), 340 audioBuffer.i16, audioBuffer.size); 341 buffer->set_range(0, bufferSize); 342 queueInputBuffer_l(buffer, timeUs); 343 return OK; 344} 345 346void AudioSource::queueInputBuffer_l(MediaBuffer *buffer, int64_t timeUs) { 347 const size_t bufferSize = buffer->range_length(); 348 const size_t frameSize = mRecord->frameSize(); 349 const int64_t timestampUs = 350 mPrevSampleTimeUs + 351 ((1000000LL * (bufferSize / frameSize)) + 352 (mSampleRate >> 1)) / mSampleRate; 353 354 if (mNumFramesReceived == 0) { 355 buffer->meta_data()->setInt64(kKeyAnchorTime, mStartTimeUs); 356 } 357 358 buffer->meta_data()->setInt64(kKeyTime, mPrevSampleTimeUs); 359 buffer->meta_data()->setInt64(kKeyDriftTime, timeUs - mInitialReadTimeUs); 360 mPrevSampleTimeUs = timestampUs; 361 mNumFramesReceived += bufferSize / frameSize; 362 mBuffersReceived.push_back(buffer); 363 mFrameAvailableCondition.signal(); 364} 365 366void AudioSource::trackMaxAmplitude(int16_t *data, int nSamples) { 367 for (int i = nSamples; i > 0; --i) { 368 int16_t value = *data++; 369 if (value < 0) { 370 value = -value; 371 } 372 if (mMaxAmplitude < value) { 373 mMaxAmplitude = value; 374 } 375 } 376} 377 378int16_t AudioSource::getMaxAmplitude() { 379 // First call activates the tracking. 380 if (!mTrackMaxAmplitude) { 381 mTrackMaxAmplitude = true; 382 } 383 int16_t value = mMaxAmplitude; 384 mMaxAmplitude = 0; 385 ALOGV("max amplitude since last call: %d", value); 386 return value; 387} 388 389} // namespace android 390