AudioSource.cpp revision a5750e0dad9e90f2195ce36f2c4457fa04b2b83e 
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    mRecord->stop();
159    waitOutstandingEncodingFrames_l();
160    releaseQueuedFrames_l();
161
162    return OK;
163}
164
165sp<MetaData> AudioSource::getFormat() {
166    Mutex::Autolock autoLock(mLock);
167    if (mInitCheck != OK) {
168        return 0;
169    }
170
171    sp<MetaData> meta = new MetaData;
172    meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_RAW);
173    meta->setInt32(kKeySampleRate, mSampleRate);
174    meta->setInt32(kKeyChannelCount, mRecord->channelCount());
175    meta->setInt32(kKeyMaxInputSize, kMaxBufferSize);
176
177    return meta;
178}
179
180void AudioSource::rampVolume(
181        int32_t startFrame, int32_t rampDurationFrames,
182        uint8_t *data,   size_t bytes) {
183
184    const int32_t kShift = 14;
185    int32_t fixedMultiplier = (startFrame << kShift) / rampDurationFrames;
186    const int32_t nChannels = mRecord->channelCount();
187    int32_t stopFrame = startFrame + bytes / sizeof(int16_t);
188    int16_t *frame = (int16_t *) data;
189    if (stopFrame > rampDurationFrames) {
190        stopFrame = rampDurationFrames;
191    }
192
193    while (startFrame < stopFrame) {
194        if (nChannels == 1) {  // mono
195            frame[0] = (frame[0] * fixedMultiplier) >> kShift;
196            ++frame;
197            ++startFrame;
198        } else {               // stereo
199            frame[0] = (frame[0] * fixedMultiplier) >> kShift;
200            frame[1] = (frame[1] * fixedMultiplier) >> kShift;
201            frame += 2;
202            startFrame += 2;
203        }
204
205        // Update the multiplier every 4 frames
206        if ((startFrame & 3) == 0) {
207            fixedMultiplier = (startFrame << kShift) / rampDurationFrames;
208        }
209    }
210}
211
212status_t AudioSource::read(
213        MediaBuffer **out, const ReadOptions * /* options */) {
214    Mutex::Autolock autoLock(mLock);
215    *out = NULL;
216
217    if (mInitCheck != OK) {
218        return NO_INIT;
219    }
220
221    while (mStarted && mBuffersReceived.empty()) {
222        mFrameAvailableCondition.wait(mLock);
223    }
224    if (!mStarted) {
225        return OK;
226    }
227    MediaBuffer *buffer = *mBuffersReceived.begin();
228    mBuffersReceived.erase(mBuffersReceived.begin());
229    ++mNumClientOwnedBuffers;
230    buffer->setObserver(this);
231    buffer->add_ref();
232
233    // Mute/suppress the recording sound
234    int64_t timeUs;
235    CHECK(buffer->meta_data()->findInt64(kKeyTime, &timeUs));
236    int64_t elapsedTimeUs = timeUs - mStartTimeUs;
237    if (elapsedTimeUs < kAutoRampStartUs) {
238        memset((uint8_t *) buffer->data(), 0, buffer->range_length());
239    } else if (elapsedTimeUs < kAutoRampStartUs + kAutoRampDurationUs) {
240        int32_t autoRampDurationFrames =
241                    ((int64_t)kAutoRampDurationUs * mSampleRate + 500000LL) / 1000000LL; //Need type casting
242
243        int32_t autoRampStartFrames =
244                    ((int64_t)kAutoRampStartUs * mSampleRate + 500000LL) / 1000000LL; //Need type casting
245
246        int32_t nFrames = mNumFramesReceived - autoRampStartFrames;
247        rampVolume(nFrames, autoRampDurationFrames,
248                (uint8_t *) buffer->data(), buffer->range_length());
249    }
250
251    // Track the max recording signal amplitude.
252    if (mTrackMaxAmplitude) {
253        trackMaxAmplitude(
254            (int16_t *) buffer->data(), buffer->range_length() >> 1);
255    }
256
257    *out = buffer;
258    return OK;
259}
260
261void AudioSource::signalBufferReturned(MediaBuffer *buffer) {
262    ALOGV("signalBufferReturned: %p", buffer->data());
263    Mutex::Autolock autoLock(mLock);
264    --mNumClientOwnedBuffers;
265    buffer->setObserver(0);
266    buffer->release();
267    mFrameEncodingCompletionCondition.signal();
268    return;
269}
270
271status_t AudioSource::dataCallback(const AudioRecord::Buffer& audioBuffer) {
272    int64_t timeUs = systemTime() / 1000ll;
273
274    ALOGV("dataCallbackTimestamp: %" PRId64 " us", timeUs);
275    Mutex::Autolock autoLock(mLock);
276    if (!mStarted) {
277        ALOGW("Spurious callback from AudioRecord. Drop the audio data.");
278        return OK;
279    }
280
281    // Drop retrieved and previously lost audio data.
282    if (mNumFramesReceived == 0 && timeUs < mStartTimeUs) {
283        (void) mRecord->getInputFramesLost();
284        ALOGV("Drop audio data at %" PRId64 "/%" PRId64 " us", timeUs, mStartTimeUs);
285        return OK;
286    }
287
288    if (mNumFramesReceived == 0 && mPrevSampleTimeUs == 0) {
289        mInitialReadTimeUs = timeUs;
290        // Initial delay
291        if (mStartTimeUs > 0) {
292            mStartTimeUs = timeUs - mStartTimeUs;
293        } else {
294            // Assume latency is constant.
295            mStartTimeUs += mRecord->latency() * 1000;
296        }
297
298        mPrevSampleTimeUs = mStartTimeUs;
299    }
300
301    size_t numLostBytes = 0;
302    if (mNumFramesReceived > 0) {  // Ignore earlier frame lost
303        // getInputFramesLost() returns the number of lost frames.
304        // Convert number of frames lost to number of bytes lost.
305        numLostBytes = mRecord->getInputFramesLost() * mRecord->frameSize();
306    }
307
308    CHECK_EQ(numLostBytes & 1, 0u);
309    CHECK_EQ(audioBuffer.size & 1, 0u);
310    if (numLostBytes > 0) {
311        // Loss of audio frames should happen rarely; thus the LOGW should
312        // not cause a logging spam
313        ALOGW("Lost audio record data: %zu bytes", numLostBytes);
314    }
315
316    while (numLostBytes > 0) {
317        size_t bufferSize = numLostBytes;
318        if (numLostBytes > kMaxBufferSize) {
319            numLostBytes -= kMaxBufferSize;
320            bufferSize = kMaxBufferSize;
321        } else {
322            numLostBytes = 0;
323        }
324        MediaBuffer *lostAudioBuffer = new MediaBuffer(bufferSize);
325        memset(lostAudioBuffer->data(), 0, bufferSize);
326        lostAudioBuffer->set_range(0, bufferSize);
327        queueInputBuffer_l(lostAudioBuffer, timeUs);
328    }
329
330    if (audioBuffer.size == 0) {
331        ALOGW("Nothing is available from AudioRecord callback buffer");
332        return OK;
333    }
334
335    const size_t bufferSize = audioBuffer.size;
336    MediaBuffer *buffer = new MediaBuffer(bufferSize);
337    memcpy((uint8_t *) buffer->data(),
338            audioBuffer.i16, audioBuffer.size);
339    buffer->set_range(0, bufferSize);
340    queueInputBuffer_l(buffer, timeUs);
341    return OK;
342}
343
344void AudioSource::queueInputBuffer_l(MediaBuffer *buffer, int64_t timeUs) {
345    const size_t bufferSize = buffer->range_length();
346    const size_t frameSize = mRecord->frameSize();
347    const int64_t timestampUs =
348                mPrevSampleTimeUs +
349                    ((1000000LL * (bufferSize / frameSize)) +
350                        (mSampleRate >> 1)) / mSampleRate;
351
352    if (mNumFramesReceived == 0) {
353        buffer->meta_data()->setInt64(kKeyAnchorTime, mStartTimeUs);
354    }
355
356    buffer->meta_data()->setInt64(kKeyTime, mPrevSampleTimeUs);
357    buffer->meta_data()->setInt64(kKeyDriftTime, timeUs - mInitialReadTimeUs);
358    mPrevSampleTimeUs = timestampUs;
359    mNumFramesReceived += bufferSize / frameSize;
360    mBuffersReceived.push_back(buffer);
361    mFrameAvailableCondition.signal();
362}
363
364void AudioSource::trackMaxAmplitude(int16_t *data, int nSamples) {
365    for (int i = nSamples; i > 0; --i) {
366        int16_t value = *data++;
367        if (value < 0) {
368            value = -value;
369        }
370        if (mMaxAmplitude < value) {
371            mMaxAmplitude = value;
372        }
373    }
374}
375
376int16_t AudioSource::getMaxAmplitude() {
377    // First call activates the tracking.
378    if (!mTrackMaxAmplitude) {
379        mTrackMaxAmplitude = true;
380    }
381    int16_t value = mMaxAmplitude;
382    mMaxAmplitude = 0;
383    ALOGV("max amplitude since last call: %d", value);
384    return value;
385}
386
387}  // namespace android
388