AudioSource.cpp revision eaae38445a340c4857c1c5569475879a728e63b7
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 if (err == OK) { 95 mGroup = new MediaBufferGroup; 96 mGroup->add_buffer(new MediaBuffer(kMaxBufferSize)); 97 98 mStarted = true; 99 } else { 100 delete mRecord; 101 mRecord = NULL; 102 } 103 104 105 return err; 106} 107 108status_t AudioSource::stop() { 109 if (!mStarted) { 110 return UNKNOWN_ERROR; 111 } 112 113 if (mInitCheck != OK) { 114 return NO_INIT; 115 } 116 117 mRecord->stop(); 118 119 delete mGroup; 120 mGroup = NULL; 121 122 mStarted = false; 123 124 if (mCollectStats) { 125 LOGI("Total lost audio frames: %lld", 126 mTotalLostFrames + (mPrevLostBytes >> 1)); 127 } 128 129 return OK; 130} 131 132sp<MetaData> AudioSource::getFormat() { 133 if (mInitCheck != OK) { 134 return 0; 135 } 136 137 sp<MetaData> meta = new MetaData; 138 meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_RAW); 139 meta->setInt32(kKeySampleRate, mRecord->getSampleRate()); 140 meta->setInt32(kKeyChannelCount, mRecord->channelCount()); 141 meta->setInt32(kKeyMaxInputSize, kMaxBufferSize); 142 143 return meta; 144} 145 146void AudioSource::rampVolume( 147 int32_t startFrame, int32_t rampDurationFrames, 148 uint8_t *data, size_t bytes) { 149 150 const int32_t kShift = 14; 151 int32_t fixedMultiplier = (startFrame << kShift) / rampDurationFrames; 152 const int32_t nChannels = mRecord->channelCount(); 153 int32_t stopFrame = startFrame + bytes / sizeof(int16_t); 154 int16_t *frame = (int16_t *) data; 155 if (stopFrame > rampDurationFrames) { 156 stopFrame = rampDurationFrames; 157 } 158 159 while (startFrame < stopFrame) { 160 if (nChannels == 1) { // mono 161 frame[0] = (frame[0] * fixedMultiplier) >> kShift; 162 ++frame; 163 ++startFrame; 164 } else { // stereo 165 frame[0] = (frame[0] * fixedMultiplier) >> kShift; 166 frame[1] = (frame[1] * fixedMultiplier) >> kShift; 167 frame += 2; 168 startFrame += 2; 169 } 170 171 // Update the multiplier every 4 frames 172 if ((startFrame & 3) == 0) { 173 fixedMultiplier = (startFrame << kShift) / rampDurationFrames; 174 } 175 } 176} 177 178status_t AudioSource::read( 179 MediaBuffer **out, const ReadOptions *options) { 180 181 if (mInitCheck != OK) { 182 return NO_INIT; 183 } 184 185 int64_t readTimeUs = systemTime() / 1000; 186 *out = NULL; 187 188 MediaBuffer *buffer; 189 CHECK_EQ(mGroup->acquire_buffer(&buffer), OK); 190 191 int err = 0; 192 if (mStarted) { 193 194 uint32_t numFramesRecorded; 195 mRecord->getPosition(&numFramesRecorded); 196 197 198 if (numFramesRecorded == 0 && mPrevSampleTimeUs == 0) { 199 mInitialReadTimeUs = readTimeUs; 200 // Initial delay 201 if (mStartTimeUs > 0) { 202 mStartTimeUs = readTimeUs - mStartTimeUs; 203 } else { 204 // Assume latency is constant. 205 mStartTimeUs += mRecord->latency() * 1000; 206 } 207 mPrevSampleTimeUs = mStartTimeUs; 208 } 209 210 uint32_t sampleRate = mRecord->getSampleRate(); 211 212 // Insert null frames when lost frames are detected. 213 int64_t timestampUs = mPrevSampleTimeUs; 214 uint32_t numLostBytes = mRecord->getInputFramesLost() << 1; 215 numLostBytes += mPrevLostBytes; 216#if 0 217 // Simulate lost frames 218 numLostBytes = ((rand() * 1.0 / RAND_MAX)) * 2 * kMaxBufferSize; 219 numLostBytes &= 0xFFFFFFFE; // Alignment requirement 220 221 // Reduce the chance to lose 222 if (rand() * 1.0 / RAND_MAX >= 0.05) { 223 numLostBytes = 0; 224 } 225#endif 226 if (numLostBytes > 0) { 227 if (numLostBytes > kMaxBufferSize) { 228 mPrevLostBytes = numLostBytes - kMaxBufferSize; 229 numLostBytes = kMaxBufferSize; 230 } else { 231 mPrevLostBytes = 0; 232 } 233 234 CHECK_EQ(numLostBytes & 1, 0); 235 timestampUs += ((1000000LL * (numLostBytes >> 1)) + 236 (sampleRate >> 1)) / sampleRate; 237 238 CHECK(timestampUs > mPrevSampleTimeUs); 239 if (mCollectStats) { 240 mTotalLostFrames += (numLostBytes >> 1); 241 } 242 memset(buffer->data(), 0, numLostBytes); 243 buffer->set_range(0, numLostBytes); 244 if (numFramesRecorded == 0) { 245 buffer->meta_data()->setInt64(kKeyAnchorTime, mStartTimeUs); 246 } 247 buffer->meta_data()->setInt64(kKeyTime, mStartTimeUs + mPrevSampleTimeUs); 248 buffer->meta_data()->setInt64(kKeyDriftTime, readTimeUs - mInitialReadTimeUs); 249 mPrevSampleTimeUs = timestampUs; 250 *out = buffer; 251 return OK; 252 } 253 254 ssize_t n = mRecord->read(buffer->data(), buffer->size()); 255 if (n < 0) { 256 buffer->release(); 257 return (status_t)n; 258 } 259 260 int64_t recordDurationUs = (1000000LL * n >> 1) / sampleRate; 261 timestampUs += recordDurationUs; 262 263 if (mPrevSampleTimeUs - mStartTimeUs < kAutoRampStartUs) { 264 // Mute the initial video recording signal 265 memset((uint8_t *) buffer->data(), 0, n); 266 } else if (mPrevSampleTimeUs - mStartTimeUs < kAutoRampStartUs + kAutoRampDurationUs) { 267 int32_t autoRampDurationFrames = 268 (kAutoRampDurationUs * sampleRate + 500000LL) / 1000000LL; 269 270 int32_t autoRampStartFrames = 271 (kAutoRampStartUs * sampleRate + 500000LL) / 1000000LL; 272 273 int32_t nFrames = numFramesRecorded - autoRampStartFrames; 274 rampVolume(nFrames, autoRampDurationFrames, (uint8_t *) buffer->data(), n); 275 } 276 if (mTrackMaxAmplitude) { 277 trackMaxAmplitude((int16_t *) buffer->data(), n >> 1); 278 } 279 280 if (numFramesRecorded == 0) { 281 buffer->meta_data()->setInt64(kKeyAnchorTime, mStartTimeUs); 282 } 283 284 buffer->meta_data()->setInt64(kKeyTime, mStartTimeUs + mPrevSampleTimeUs); 285 buffer->meta_data()->setInt64(kKeyDriftTime, readTimeUs - mInitialReadTimeUs); 286 CHECK(timestampUs > mPrevSampleTimeUs); 287 mPrevSampleTimeUs = timestampUs; 288 LOGV("initial delay: %lld, sample rate: %d, timestamp: %lld", 289 mStartTimeUs, sampleRate, timestampUs); 290 291 buffer->set_range(0, n); 292 293 *out = buffer; 294 return OK; 295 } 296 297 return OK; 298} 299 300void AudioSource::trackMaxAmplitude(int16_t *data, int nSamples) { 301 for (int i = nSamples; i > 0; --i) { 302 int16_t value = *data++; 303 if (value < 0) { 304 value = -value; 305 } 306 if (mMaxAmplitude < value) { 307 mMaxAmplitude = value; 308 } 309 } 310} 311 312int16_t AudioSource::getMaxAmplitude() { 313 // First call activates the tracking. 314 if (!mTrackMaxAmplitude) { 315 mTrackMaxAmplitude = true; 316 } 317 int16_t value = mMaxAmplitude; 318 mMaxAmplitude = 0; 319 LOGV("max amplitude since last call: %d", value); 320 return value; 321} 322 323} // namespace android 324