AudioSource.cpp revision 679ab0b0792846a89162ce41c953819d70030112 
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/AudioRecord.h>
22#include <media/stagefright/AudioSource.h>
23#include <media/stagefright/MediaBuffer.h>
24#include <media/stagefright/MediaDefs.h>
25#include <media/stagefright/MetaData.h>
26#include <media/stagefright/foundation/ADebug.h>
27#include <cutils/properties.h>
28#include <stdlib.h>
29
30namespace android {
31
32static void AudioRecordCallbackFunction(int event, void *user, void *info) {
33    AudioSource *source = (AudioSource *) user;
34    switch (event) {
35        case AudioRecord::EVENT_MORE_DATA: {
36            source->dataCallbackTimestamp(*((AudioRecord::Buffer *) info), systemTime() / 1000);
37            break;
38        }
39        case AudioRecord::EVENT_OVERRUN: {
40            ALOGW("AudioRecord reported overrun!");
41            break;
42        }
43        default:
44            // does nothing
45            break;
46    }
47}
48
49AudioSource::AudioSource(
50        audio_source_t inputSource, uint32_t sampleRate, uint32_t channels)
51    : mStarted(false),
52      mSampleRate(sampleRate),
53      mPrevSampleTimeUs(0),
54      mNumFramesReceived(0),
55      mNumClientOwnedBuffers(0) {
56
57    ALOGV("sampleRate: %d, channels: %d", sampleRate, channels);
58    CHECK(channels == 1 || channels == 2);
59    AudioRecord::record_flags flags = (AudioRecord::record_flags)
60                    (AudioRecord::RECORD_AGC_ENABLE |
61                     AudioRecord::RECORD_NS_ENABLE  |
62                     AudioRecord::RECORD_IIR_ENABLE);
63    mRecord = new AudioRecord(
64                inputSource, sampleRate, AUDIO_FORMAT_PCM_16_BIT,
65                channels > 1? AUDIO_CHANNEL_IN_STEREO: AUDIO_CHANNEL_IN_MONO,
66                4 * kMaxBufferSize / sizeof(int16_t), /* Enable ping-pong buffers */
67                flags,
68                AudioRecordCallbackFunction,
69                this);
70
71    mInitCheck = mRecord->initCheck();
72}
73
74AudioSource::~AudioSource() {
75    if (mStarted) {
76        reset();
77    }
78
79    delete mRecord;
80    mRecord = NULL;
81}
82
83status_t AudioSource::initCheck() const {
84    return mInitCheck;
85}
86
87status_t AudioSource::start(MetaData *params) {
88    Mutex::Autolock autoLock(mLock);
89    if (mStarted) {
90        return UNKNOWN_ERROR;
91    }
92
93    if (mInitCheck != OK) {
94        return NO_INIT;
95    }
96
97    mTrackMaxAmplitude = false;
98    mMaxAmplitude = 0;
99    mInitialReadTimeUs = 0;
100    mStartTimeUs = 0;
101    int64_t startTimeUs;
102    if (params && params->findInt64(kKeyTime, &startTimeUs)) {
103        mStartTimeUs = startTimeUs;
104    }
105    status_t err = mRecord->start();
106    if (err == OK) {
107        mStarted = true;
108    } else {
109        delete mRecord;
110        mRecord = NULL;
111    }
112
113
114    return err;
115}
116
117void AudioSource::releaseQueuedFrames_l() {
118    ALOGV("releaseQueuedFrames_l");
119    List<MediaBuffer *>::iterator it;
120    while (!mBuffersReceived.empty()) {
121        it = mBuffersReceived.begin();
122        (*it)->release();
123        mBuffersReceived.erase(it);
124    }
125}
126
127void AudioSource::waitOutstandingEncodingFrames_l() {
128    ALOGV("waitOutstandingEncodingFrames_l: %lld", mNumClientOwnedBuffers);
129    while (mNumClientOwnedBuffers > 0) {
130        mFrameEncodingCompletionCondition.wait(mLock);
131    }
132}
133
134status_t AudioSource::reset() {
135    Mutex::Autolock autoLock(mLock);
136    if (!mStarted) {
137        return UNKNOWN_ERROR;
138    }
139
140    if (mInitCheck != OK) {
141        return NO_INIT;
142    }
143
144    mStarted = false;
145    mRecord->stop();
146    waitOutstandingEncodingFrames_l();
147    releaseQueuedFrames_l();
148
149    return OK;
150}
151
152sp<MetaData> AudioSource::getFormat() {
153    Mutex::Autolock autoLock(mLock);
154    if (mInitCheck != OK) {
155        return 0;
156    }
157
158    sp<MetaData> meta = new MetaData;
159    meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_RAW);
160    meta->setInt32(kKeySampleRate, mSampleRate);
161    meta->setInt32(kKeyChannelCount, mRecord->channelCount());
162    meta->setInt32(kKeyMaxInputSize, kMaxBufferSize);
163
164    return meta;
165}
166
167void AudioSource::rampVolume(
168        int32_t startFrame, int32_t rampDurationFrames,
169        uint8_t *data,   size_t bytes) {
170
171    const int32_t kShift = 14;
172    int32_t fixedMultiplier = (startFrame << kShift) / rampDurationFrames;
173    const int32_t nChannels = mRecord->channelCount();
174    int32_t stopFrame = startFrame + bytes / sizeof(int16_t);
175    int16_t *frame = (int16_t *) data;
176    if (stopFrame > rampDurationFrames) {
177        stopFrame = rampDurationFrames;
178    }
179
180    while (startFrame < stopFrame) {
181        if (nChannels == 1) {  // mono
182            frame[0] = (frame[0] * fixedMultiplier) >> kShift;
183            ++frame;
184            ++startFrame;
185        } else {               // stereo
186            frame[0] = (frame[0] * fixedMultiplier) >> kShift;
187            frame[1] = (frame[1] * fixedMultiplier) >> kShift;
188            frame += 2;
189            startFrame += 2;
190        }
191
192        // Update the multiplier every 4 frames
193        if ((startFrame & 3) == 0) {
194            fixedMultiplier = (startFrame << kShift) / rampDurationFrames;
195        }
196    }
197}
198
199status_t AudioSource::read(
200        MediaBuffer **out, const ReadOptions *options) {
201    Mutex::Autolock autoLock(mLock);
202    *out = NULL;
203
204    if (mInitCheck != OK) {
205        return NO_INIT;
206    }
207
208    while (mStarted && mBuffersReceived.empty()) {
209        mFrameAvailableCondition.wait(mLock);
210    }
211    if (!mStarted) {
212        return OK;
213    }
214    MediaBuffer *buffer = *mBuffersReceived.begin();
215    mBuffersReceived.erase(mBuffersReceived.begin());
216    ++mNumClientOwnedBuffers;
217    buffer->setObserver(this);
218    buffer->add_ref();
219
220    // Mute/suppress the recording sound
221    int64_t timeUs;
222    CHECK(buffer->meta_data()->findInt64(kKeyTime, &timeUs));
223    int64_t elapsedTimeUs = timeUs - mStartTimeUs;
224    if (elapsedTimeUs < kAutoRampStartUs) {
225        memset((uint8_t *) buffer->data(), 0, buffer->range_length());
226    } else if (elapsedTimeUs < kAutoRampStartUs + kAutoRampDurationUs) {
227        int32_t autoRampDurationFrames =
228                    (kAutoRampDurationUs * mSampleRate + 500000LL) / 1000000LL;
229
230        int32_t autoRampStartFrames =
231                    (kAutoRampStartUs * mSampleRate + 500000LL) / 1000000LL;
232
233        int32_t nFrames = mNumFramesReceived - autoRampStartFrames;
234        rampVolume(nFrames, autoRampDurationFrames,
235                (uint8_t *) buffer->data(), buffer->range_length());
236    }
237
238    // Track the max recording signal amplitude.
239    if (mTrackMaxAmplitude) {
240        trackMaxAmplitude(
241            (int16_t *) buffer->data(), buffer->range_length() >> 1);
242    }
243
244    *out = buffer;
245    return OK;
246}
247
248void AudioSource::signalBufferReturned(MediaBuffer *buffer) {
249    ALOGV("signalBufferReturned: %p", buffer->data());
250    Mutex::Autolock autoLock(mLock);
251    --mNumClientOwnedBuffers;
252    buffer->setObserver(0);
253    buffer->release();
254    mFrameEncodingCompletionCondition.signal();
255    return;
256}
257
258status_t AudioSource::dataCallbackTimestamp(
259        const AudioRecord::Buffer& audioBuffer, int64_t timeUs) {
260    ALOGV("dataCallbackTimestamp: %lld us", timeUs);
261    Mutex::Autolock autoLock(mLock);
262    if (!mStarted) {
263        ALOGW("Spurious callback from AudioRecord. Drop the audio data.");
264        return OK;
265    }
266
267    // Drop retrieved and previously lost audio data.
268    if (mNumFramesReceived == 0 && timeUs < mStartTimeUs) {
269        mRecord->getInputFramesLost();
270        ALOGV("Drop audio data at %lld/%lld us", timeUs, mStartTimeUs);
271        return OK;
272    }
273
274    if (mNumFramesReceived == 0 && mPrevSampleTimeUs == 0) {
275        mInitialReadTimeUs = timeUs;
276        // Initial delay
277        if (mStartTimeUs > 0) {
278            mStartTimeUs = timeUs - mStartTimeUs;
279        } else {
280            // Assume latency is constant.
281            mStartTimeUs += mRecord->latency() * 1000;
282        }
283        mPrevSampleTimeUs = mStartTimeUs;
284    }
285
286    size_t numLostBytes = 0;
287    if (mNumFramesReceived > 0) {  // Ignore earlier frame lost
288        // getInputFramesLost() returns the number of lost frames.
289        // Convert number of frames lost to number of bytes lost.
290        numLostBytes = mRecord->getInputFramesLost() * mRecord->frameSize();
291    }
292
293    CHECK_EQ(numLostBytes & 1, 0u);
294    CHECK_EQ(audioBuffer.size & 1, 0u);
295    if (numLostBytes > 0) {
296        // Loss of audio frames should happen rarely; thus the LOGW should
297        // not cause a logging spam
298        ALOGW("Lost audio record data: %d bytes", numLostBytes);
299    }
300
301    while (numLostBytes > 0) {
302        size_t bufferSize = numLostBytes;
303        if (numLostBytes > kMaxBufferSize) {
304            numLostBytes -= kMaxBufferSize;
305            bufferSize = kMaxBufferSize;
306        } else {
307            numLostBytes = 0;
308        }
309        MediaBuffer *lostAudioBuffer = new MediaBuffer(bufferSize);
310        memset(lostAudioBuffer->data(), 0, bufferSize);
311        lostAudioBuffer->set_range(0, bufferSize);
312        queueInputBuffer_l(lostAudioBuffer, timeUs);
313    }
314
315    if (audioBuffer.size == 0) {
316        ALOGW("Nothing is available from AudioRecord callback buffer");
317        return OK;
318    }
319
320    const size_t bufferSize = audioBuffer.size;
321    MediaBuffer *buffer = new MediaBuffer(bufferSize);
322    memcpy((uint8_t *) buffer->data(),
323            audioBuffer.i16, audioBuffer.size);
324    buffer->set_range(0, bufferSize);
325    queueInputBuffer_l(buffer, timeUs);
326    return OK;
327}
328
329void AudioSource::queueInputBuffer_l(MediaBuffer *buffer, int64_t timeUs) {
330    const size_t bufferSize = buffer->range_length();
331    const size_t frameSize = mRecord->frameSize();
332    const int64_t timestampUs =
333                mPrevSampleTimeUs +
334                    ((1000000LL * (bufferSize / frameSize)) +
335                        (mSampleRate >> 1)) / mSampleRate;
336
337    if (mNumFramesReceived == 0) {
338        buffer->meta_data()->setInt64(kKeyAnchorTime, mStartTimeUs);
339    }
340
341    buffer->meta_data()->setInt64(kKeyTime, mPrevSampleTimeUs);
342    buffer->meta_data()->setInt64(kKeyDriftTime, timeUs - mInitialReadTimeUs);
343    mPrevSampleTimeUs = timestampUs;
344    mNumFramesReceived += bufferSize / frameSize;
345    mBuffersReceived.push_back(buffer);
346    mFrameAvailableCondition.signal();
347}
348
349void AudioSource::trackMaxAmplitude(int16_t *data, int nSamples) {
350    for (int i = nSamples; i > 0; --i) {
351        int16_t value = *data++;
352        if (value < 0) {
353            value = -value;
354        }
355        if (mMaxAmplitude < value) {
356            mMaxAmplitude = value;
357        }
358    }
359}
360
361int16_t AudioSource::getMaxAmplitude() {
362    // First call activates the tracking.
363    if (!mTrackMaxAmplitude) {
364        mTrackMaxAmplitude = true;
365    }
366    int16_t value = mMaxAmplitude;
367    mMaxAmplitude = 0;
368    ALOGV("max amplitude since last call: %d", value);
369    return value;
370}
371
372}  // namespace android
373