AudioSource.cpp revision 3c3763d2ee1cd1fba7fe522fbaf0faca315d8c2a
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//#define LOG_NDEBUG 0 18#define LOG_TAG "AudioSource" 19#include <utils/Log.h> 20 21#include <media/stagefright/AudioSource.h> 22 23#include <media/AudioRecord.h> 24#include <media/stagefright/MediaBufferGroup.h> 25#include <media/stagefright/MediaDebug.h> 26#include <media/stagefright/MediaDefs.h> 27#include <media/stagefright/MetaData.h> 28#include <cutils/properties.h> 29#include <stdlib.h> 30 31namespace android { 32 33AudioSource::AudioSource( 34 int inputSource, uint32_t sampleRate, uint32_t channels) 35 : mStarted(false), 36 mCollectStats(false), 37 mPrevSampleTimeUs(0), 38 mTotalLostFrames(0), 39 mPrevLostBytes(0), 40 mGroup(NULL) { 41 42 LOGV("sampleRate: %d, channels: %d", sampleRate, channels); 43 CHECK(channels == 1 || channels == 2); 44 uint32_t flags = AudioRecord::RECORD_AGC_ENABLE | 45 AudioRecord::RECORD_NS_ENABLE | 46 AudioRecord::RECORD_IIR_ENABLE; 47 48 mRecord = new AudioRecord( 49 inputSource, sampleRate, AudioSystem::PCM_16_BIT, 50 channels > 1? AudioSystem::CHANNEL_IN_STEREO: AudioSystem::CHANNEL_IN_MONO, 51 4 * kMaxBufferSize / sizeof(int16_t), /* Enable ping-pong buffers */ 52 flags); 53 54 mInitCheck = mRecord->initCheck(); 55} 56 57AudioSource::~AudioSource() { 58 if (mStarted) { 59 stop(); 60 } 61 62 delete mRecord; 63 mRecord = NULL; 64} 65 66status_t AudioSource::initCheck() const { 67 return mInitCheck; 68} 69 70status_t AudioSource::start(MetaData *params) { 71 if (mStarted) { 72 return UNKNOWN_ERROR; 73 } 74 75 if (mInitCheck != OK) { 76 return NO_INIT; 77 } 78 79 char value[PROPERTY_VALUE_MAX]; 80 if (property_get("media.stagefright.record-stats", value, NULL) 81 && (!strcmp(value, "1") || !strcasecmp(value, "true"))) { 82 mCollectStats = true; 83 } 84 85 mTrackMaxAmplitude = false; 86 mMaxAmplitude = 0; 87 mInitialReadTimeUs = 0; 88 mStartTimeUs = 0; 89 int64_t startTimeUs; 90 if (params && params->findInt64(kKeyTime, &startTimeUs)) { 91 mStartTimeUs = startTimeUs; 92 } 93 status_t err = mRecord->start(); 94 95 if (err == OK) { 96 mGroup = new MediaBufferGroup; 97 mGroup->add_buffer(new MediaBuffer(kMaxBufferSize)); 98 99 mStarted = true; 100 } 101 102 return err; 103} 104 105status_t AudioSource::stop() { 106 if (!mStarted) { 107 return UNKNOWN_ERROR; 108 } 109 110 if (mInitCheck != OK) { 111 return NO_INIT; 112 } 113 114 mRecord->stop(); 115 116 delete mGroup; 117 mGroup = NULL; 118 119 mStarted = false; 120 121 if (mCollectStats) { 122 LOGI("Total lost audio frames: %lld", 123 mTotalLostFrames + (mPrevLostBytes >> 1)); 124 } 125 126 return OK; 127} 128 129sp<MetaData> AudioSource::getFormat() { 130 if (mInitCheck != OK) { 131 return 0; 132 } 133 134 sp<MetaData> meta = new MetaData; 135 meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_RAW); 136 meta->setInt32(kKeySampleRate, mRecord->getSampleRate()); 137 meta->setInt32(kKeyChannelCount, mRecord->channelCount()); 138 meta->setInt32(kKeyMaxInputSize, kMaxBufferSize); 139 140 return meta; 141} 142 143/* 144 * Returns -1 if frame skipping request is too long. 145 * Returns 0 if there is no need to skip frames. 146 * Returns 1 if we need to skip frames. 147 */ 148static int skipFrame(int64_t timestampUs, 149 const MediaSource::ReadOptions *options) { 150 151 int64_t skipFrameUs; 152 if (!options || !options->getSkipFrame(&skipFrameUs)) { 153 return 0; 154 } 155 156 if (skipFrameUs <= timestampUs) { 157 return 0; 158 } 159 160 // Safe guard against the abuse of the kSkipFrame_Option. 161 if (skipFrameUs - timestampUs >= 1E6) { 162 LOGE("Frame skipping requested is way too long: %lld us", 163 skipFrameUs - timestampUs); 164 165 return -1; 166 } 167 168 LOGV("skipFrame: %lld us > timestamp: %lld us", 169 skipFrameUs, timestampUs); 170 171 return 1; 172 173} 174 175void AudioSource::rampVolume( 176 int32_t startFrame, int32_t rampDurationFrames, 177 uint8_t *data, size_t bytes) { 178 179 const int32_t kShift = 14; 180 int32_t fixedMultiplier = (startFrame << kShift) / rampDurationFrames; 181 const int32_t nChannels = mRecord->channelCount(); 182 int32_t stopFrame = startFrame + bytes / sizeof(int16_t); 183 int16_t *frame = (int16_t *) data; 184 if (stopFrame > rampDurationFrames) { 185 stopFrame = rampDurationFrames; 186 } 187 188 while (startFrame < stopFrame) { 189 if (nChannels == 1) { // mono 190 frame[0] = (frame[0] * fixedMultiplier) >> kShift; 191 ++frame; 192 ++startFrame; 193 } else { // stereo 194 frame[0] = (frame[0] * fixedMultiplier) >> kShift; 195 frame[1] = (frame[1] * fixedMultiplier) >> kShift; 196 frame += 2; 197 startFrame += 2; 198 } 199 200 // Update the multiplier every 4 frames 201 if ((startFrame & 3) == 0) { 202 fixedMultiplier = (startFrame << kShift) / rampDurationFrames; 203 } 204 } 205} 206 207status_t AudioSource::read( 208 MediaBuffer **out, const ReadOptions *options) { 209 210 if (mInitCheck != OK) { 211 return NO_INIT; 212 } 213 214 int64_t readTimeUs = systemTime() / 1000; 215 *out = NULL; 216 217 MediaBuffer *buffer; 218 CHECK_EQ(mGroup->acquire_buffer(&buffer), OK); 219 220 int err = 0; 221 while (mStarted) { 222 223 uint32_t numFramesRecorded; 224 mRecord->getPosition(&numFramesRecorded); 225 226 227 if (numFramesRecorded == 0 && mPrevSampleTimeUs == 0) { 228 mInitialReadTimeUs = readTimeUs; 229 // Initial delay 230 if (mStartTimeUs > 0) { 231 mStartTimeUs = readTimeUs - mStartTimeUs; 232 } else { 233 // Assume latency is constant. 234 mStartTimeUs += mRecord->latency() * 1000; 235 } 236 mPrevSampleTimeUs = mStartTimeUs; 237 } 238 239 uint32_t sampleRate = mRecord->getSampleRate(); 240 241 // Insert null frames when lost frames are detected. 242 int64_t timestampUs = mPrevSampleTimeUs; 243 uint32_t numLostBytes = mRecord->getInputFramesLost() << 1; 244 numLostBytes += mPrevLostBytes; 245#if 0 246 // Simulate lost frames 247 numLostBytes = ((rand() * 1.0 / RAND_MAX)) * 2 * kMaxBufferSize; 248 numLostBytes &= 0xFFFFFFFE; // Alignment requirement 249 250 // Reduce the chance to lose 251 if (rand() * 1.0 / RAND_MAX >= 0.05) { 252 numLostBytes = 0; 253 } 254#endif 255 if (numLostBytes > 0) { 256 if (numLostBytes > kMaxBufferSize) { 257 mPrevLostBytes = numLostBytes - kMaxBufferSize; 258 numLostBytes = kMaxBufferSize; 259 } 260 261 CHECK_EQ(numLostBytes & 1, 0); 262 timestampUs += ((1000000LL * (numLostBytes >> 1)) + 263 (sampleRate >> 1)) / sampleRate; 264 265 CHECK(timestampUs > mPrevSampleTimeUs); 266 if (mCollectStats) { 267 mTotalLostFrames += (numLostBytes >> 1); 268 } 269 if ((err = skipFrame(timestampUs, options)) == -1) { 270 buffer->release(); 271 return UNKNOWN_ERROR; 272 } else if (err != 0) { 273 continue; 274 } 275 memset(buffer->data(), 0, numLostBytes); 276 buffer->set_range(0, numLostBytes); 277 if (numFramesRecorded == 0) { 278 buffer->meta_data()->setInt64(kKeyTime, mStartTimeUs); 279 } 280 buffer->meta_data()->setInt64(kKeyDriftTime, readTimeUs - mInitialReadTimeUs); 281 mPrevSampleTimeUs = timestampUs; 282 *out = buffer; 283 return OK; 284 } 285 286 ssize_t n = mRecord->read(buffer->data(), buffer->size()); 287 if (n < 0) { 288 buffer->release(); 289 return (status_t)n; 290 } 291 292 int64_t recordDurationUs = (1000000LL * n >> 1) / sampleRate; 293 timestampUs += recordDurationUs; 294 if ((err = skipFrame(timestampUs, options)) == -1) { 295 buffer->release(); 296 return UNKNOWN_ERROR; 297 } else if (err != 0) { 298 continue; 299 } 300 301 if (mPrevSampleTimeUs - mStartTimeUs < kAutoRampStartUs) { 302 // Mute the initial video recording signal 303 memset((uint8_t *) buffer->data(), 0, n); 304 } else if (mPrevSampleTimeUs - mStartTimeUs < kAutoRampStartUs + kAutoRampDurationUs) { 305 int32_t autoRampDurationFrames = 306 (kAutoRampDurationUs * sampleRate + 500000LL) / 1000000LL; 307 308 int32_t autoRampStartFrames = 309 (kAutoRampStartUs * sampleRate + 500000LL) / 1000000LL; 310 311 int32_t nFrames = numFramesRecorded - autoRampStartFrames; 312 rampVolume(nFrames, autoRampDurationFrames, (uint8_t *) buffer->data(), n); 313 } 314 if (mTrackMaxAmplitude) { 315 trackMaxAmplitude((int16_t *) buffer->data(), n >> 1); 316 } 317 318 if (numFramesRecorded == 0) { 319 buffer->meta_data()->setInt64(kKeyAnchorTime, mStartTimeUs); 320 } 321 322 buffer->meta_data()->setInt64(kKeyTime, mStartTimeUs + mPrevSampleTimeUs); 323 buffer->meta_data()->setInt64(kKeyDriftTime, readTimeUs - mInitialReadTimeUs); 324 CHECK(timestampUs > mPrevSampleTimeUs); 325 mPrevSampleTimeUs = timestampUs; 326 LOGV("initial delay: %lld, sample rate: %d, timestamp: %lld", 327 mStartTimeUs, sampleRate, timestampUs); 328 329 buffer->set_range(0, n); 330 331 *out = buffer; 332 return OK; 333 } 334 335 return OK; 336} 337 338void AudioSource::trackMaxAmplitude(int16_t *data, int nSamples) { 339 for (int i = nSamples; i > 0; --i) { 340 int16_t value = *data++; 341 if (value < 0) { 342 value = -value; 343 } 344 if (mMaxAmplitude < value) { 345 mMaxAmplitude = value; 346 } 347 } 348} 349 350int16_t AudioSource::getMaxAmplitude() { 351 // First call activates the tracking. 352 if (!mTrackMaxAmplitude) { 353 mTrackMaxAmplitude = true; 354 } 355 int16_t value = mMaxAmplitude; 356 mMaxAmplitude = 0; 357 LOGV("max amplitude since last call: %d", value); 358 return value; 359} 360 361} // namespace android 362