AudioSource.cpp revision 3e98ecd18c906dc3ac2ff1a890f0b3163447272d 
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, const String16 &opPackageName, uint32_t sampleRate,
54        uint32_t channelCount)
55    : mStarted(false),
56      mSampleRate(sampleRate),
57      mPrevSampleTimeUs(0),
58      mNumFramesReceived(0),
59      mNumClientOwnedBuffers(0) {
60    ALOGV("sampleRate: %d, channelCount: %d", sampleRate, channelCount);
61    CHECK(channelCount == 1 || channelCount == 2);
62
63    size_t minFrameCount;
64    status_t status = AudioRecord::getMinFrameCount(&minFrameCount,
65                                           sampleRate,
66                                           AUDIO_FORMAT_PCM_16_BIT,
67                                           audio_channel_in_mask_from_count(channelCount));
68    if (status == OK) {
69        // make sure that the AudioRecord callback never returns more than the maximum
70        // buffer size
71        uint32_t frameCount = kMaxBufferSize / sizeof(int16_t) / channelCount;
72
73        // make sure that the AudioRecord total buffer size is large enough
74        size_t bufCount = 2;
75        while ((bufCount * frameCount) < minFrameCount) {
76            bufCount++;
77        }
78
79        mRecord = new AudioRecord(
80                    inputSource, sampleRate, AUDIO_FORMAT_PCM_16_BIT,
81                    audio_channel_in_mask_from_count(channelCount),
82                    opPackageName,
83                    (size_t) (bufCount * frameCount),
84                    AudioRecordCallbackFunction,
85                    this,
86                    frameCount /*notificationFrames*/);
87        mInitCheck = mRecord->initCheck();
88        if (mInitCheck != OK) {
89            mRecord.clear();
90        }
91    } else {
92        mInitCheck = status;
93    }
94}
95
96AudioSource::~AudioSource() {
97    if (mStarted) {
98        reset();
99    }
100}
101
102status_t AudioSource::initCheck() const {
103    return mInitCheck;
104}
105
106status_t AudioSource::start(MetaData *params) {
107    Mutex::Autolock autoLock(mLock);
108    if (mStarted) {
109        return UNKNOWN_ERROR;
110    }
111
112    if (mInitCheck != OK) {
113        return NO_INIT;
114    }
115
116    mTrackMaxAmplitude = false;
117    mMaxAmplitude = 0;
118    mInitialReadTimeUs = 0;
119    mStartTimeUs = 0;
120    int64_t startTimeUs;
121    if (params && params->findInt64(kKeyTime, &startTimeUs)) {
122        mStartTimeUs = startTimeUs;
123    }
124    status_t err = mRecord->start();
125    if (err == OK) {
126        mStarted = true;
127    } else {
128        mRecord.clear();
129    }
130
131
132    return err;
133}
134
135void AudioSource::releaseQueuedFrames_l() {
136    ALOGV("releaseQueuedFrames_l");
137    List<MediaBuffer *>::iterator it;
138    while (!mBuffersReceived.empty()) {
139        it = mBuffersReceived.begin();
140        (*it)->release();
141        mBuffersReceived.erase(it);
142    }
143}
144
145void AudioSource::waitOutstandingEncodingFrames_l() {
146    ALOGV("waitOutstandingEncodingFrames_l: %" PRId64, mNumClientOwnedBuffers);
147    while (mNumClientOwnedBuffers > 0) {
148        mFrameEncodingCompletionCondition.wait(mLock);
149    }
150}
151
152status_t AudioSource::reset() {
153    Mutex::Autolock autoLock(mLock);
154    if (!mStarted) {
155        return UNKNOWN_ERROR;
156    }
157
158    if (mInitCheck != OK) {
159        return NO_INIT;
160    }
161
162    mStarted = false;
163    mFrameAvailableCondition.signal();
164
165    mRecord->stop();
166    waitOutstandingEncodingFrames_l();
167    releaseQueuedFrames_l();
168
169    return OK;
170}
171
172sp<MetaData> AudioSource::getFormat() {
173    Mutex::Autolock autoLock(mLock);
174    if (mInitCheck != OK) {
175        return 0;
176    }
177
178    sp<MetaData> meta = new MetaData;
179    meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_RAW);
180    meta->setInt32(kKeySampleRate, mSampleRate);
181    meta->setInt32(kKeyChannelCount, mRecord->channelCount());
182    meta->setInt32(kKeyMaxInputSize, kMaxBufferSize);
183
184    return meta;
185}
186
187void AudioSource::rampVolume(
188        int32_t startFrame, int32_t rampDurationFrames,
189        uint8_t *data,   size_t bytes) {
190
191    const int32_t kShift = 14;
192    int32_t fixedMultiplier = (startFrame << kShift) / rampDurationFrames;
193    const int32_t nChannels = mRecord->channelCount();
194    int32_t stopFrame = startFrame + bytes / sizeof(int16_t);
195    int16_t *frame = (int16_t *) data;
196    if (stopFrame > rampDurationFrames) {
197        stopFrame = rampDurationFrames;
198    }
199
200    while (startFrame < stopFrame) {
201        if (nChannels == 1) {  // mono
202            frame[0] = (frame[0] * fixedMultiplier) >> kShift;
203            ++frame;
204            ++startFrame;
205        } else {               // stereo
206            frame[0] = (frame[0] * fixedMultiplier) >> kShift;
207            frame[1] = (frame[1] * fixedMultiplier) >> kShift;
208            frame += 2;
209            startFrame += 2;
210        }
211
212        // Update the multiplier every 4 frames
213        if ((startFrame & 3) == 0) {
214            fixedMultiplier = (startFrame << kShift) / rampDurationFrames;
215        }
216    }
217}
218
219status_t AudioSource::read(
220        MediaBuffer **out, const ReadOptions * /* options */) {
221    Mutex::Autolock autoLock(mLock);
222    *out = NULL;
223
224    if (mInitCheck != OK) {
225        return NO_INIT;
226    }
227
228    while (mStarted && mBuffersReceived.empty()) {
229        mFrameAvailableCondition.wait(mLock);
230    }
231    if (!mStarted) {
232        return OK;
233    }
234    MediaBuffer *buffer = *mBuffersReceived.begin();
235    mBuffersReceived.erase(mBuffersReceived.begin());
236    ++mNumClientOwnedBuffers;
237    buffer->setObserver(this);
238    buffer->add_ref();
239
240    // Mute/suppress the recording sound
241    int64_t timeUs;
242    CHECK(buffer->meta_data()->findInt64(kKeyTime, &timeUs));
243    int64_t elapsedTimeUs = timeUs - mStartTimeUs;
244    if (elapsedTimeUs < kAutoRampStartUs) {
245        memset((uint8_t *) buffer->data(), 0, buffer->range_length());
246    } else if (elapsedTimeUs < kAutoRampStartUs + kAutoRampDurationUs) {
247        int32_t autoRampDurationFrames =
248                    ((int64_t)kAutoRampDurationUs * mSampleRate + 500000LL) / 1000000LL; //Need type casting
249
250        int32_t autoRampStartFrames =
251                    ((int64_t)kAutoRampStartUs * mSampleRate + 500000LL) / 1000000LL; //Need type casting
252
253        int32_t nFrames = mNumFramesReceived - autoRampStartFrames;
254        rampVolume(nFrames, autoRampDurationFrames,
255                (uint8_t *) buffer->data(), buffer->range_length());
256    }
257
258    // Track the max recording signal amplitude.
259    if (mTrackMaxAmplitude) {
260        trackMaxAmplitude(
261            (int16_t *) buffer->data(), buffer->range_length() >> 1);
262    }
263
264    *out = buffer;
265    return OK;
266}
267
268void AudioSource::signalBufferReturned(MediaBuffer *buffer) {
269    ALOGV("signalBufferReturned: %p", buffer->data());
270    Mutex::Autolock autoLock(mLock);
271    --mNumClientOwnedBuffers;
272    buffer->setObserver(0);
273    buffer->release();
274    mFrameEncodingCompletionCondition.signal();
275    return;
276}
277
278status_t AudioSource::dataCallback(const AudioRecord::Buffer& audioBuffer) {
279    int64_t timeUs = systemTime() / 1000ll;
280
281    ALOGV("dataCallbackTimestamp: %" PRId64 " us", timeUs);
282    Mutex::Autolock autoLock(mLock);
283    if (!mStarted) {
284        ALOGW("Spurious callback from AudioRecord. Drop the audio data.");
285        return OK;
286    }
287
288    // Drop retrieved and previously lost audio data.
289    if (mNumFramesReceived == 0 && timeUs < mStartTimeUs) {
290        (void) mRecord->getInputFramesLost();
291        ALOGV("Drop audio data at %" PRId64 "/%" PRId64 " us", timeUs, mStartTimeUs);
292        return OK;
293    }
294
295    if (mNumFramesReceived == 0 && mPrevSampleTimeUs == 0) {
296        mInitialReadTimeUs = timeUs;
297        // Initial delay
298        if (mStartTimeUs > 0) {
299            mStartTimeUs = timeUs - mStartTimeUs;
300        } else {
301            // Assume latency is constant.
302            mStartTimeUs += mRecord->latency() * 1000;
303        }
304
305        mPrevSampleTimeUs = mStartTimeUs;
306    }
307
308    size_t numLostBytes = 0;
309    if (mNumFramesReceived > 0) {  // Ignore earlier frame lost
310        // getInputFramesLost() returns the number of lost frames.
311        // Convert number of frames lost to number of bytes lost.
312        numLostBytes = mRecord->getInputFramesLost() * mRecord->frameSize();
313    }
314
315    CHECK_EQ(numLostBytes & 1, 0u);
316    CHECK_EQ(audioBuffer.size & 1, 0u);
317    if (numLostBytes > 0) {
318        // Loss of audio frames should happen rarely; thus the LOGW should
319        // not cause a logging spam
320        ALOGW("Lost audio record data: %zu bytes", numLostBytes);
321    }
322
323    while (numLostBytes > 0) {
324        size_t bufferSize = numLostBytes;
325        if (numLostBytes > kMaxBufferSize) {
326            numLostBytes -= kMaxBufferSize;
327            bufferSize = kMaxBufferSize;
328        } else {
329            numLostBytes = 0;
330        }
331        MediaBuffer *lostAudioBuffer = new MediaBuffer(bufferSize);
332        memset(lostAudioBuffer->data(), 0, bufferSize);
333        lostAudioBuffer->set_range(0, bufferSize);
334        queueInputBuffer_l(lostAudioBuffer, timeUs);
335    }
336
337    if (audioBuffer.size == 0) {
338        ALOGW("Nothing is available from AudioRecord callback buffer");
339        return OK;
340    }
341
342    const size_t bufferSize = audioBuffer.size;
343    MediaBuffer *buffer = new MediaBuffer(bufferSize);
344    memcpy((uint8_t *) buffer->data(),
345            audioBuffer.i16, audioBuffer.size);
346    buffer->set_range(0, bufferSize);
347    queueInputBuffer_l(buffer, timeUs);
348    return OK;
349}
350
351void AudioSource::queueInputBuffer_l(MediaBuffer *buffer, int64_t timeUs) {
352    const size_t bufferSize = buffer->range_length();
353    const size_t frameSize = mRecord->frameSize();
354    const int64_t timestampUs =
355                mPrevSampleTimeUs +
356                    ((1000000LL * (bufferSize / frameSize)) +
357                        (mSampleRate >> 1)) / mSampleRate;
358
359    if (mNumFramesReceived == 0) {
360        buffer->meta_data()->setInt64(kKeyAnchorTime, mStartTimeUs);
361    }
362
363    buffer->meta_data()->setInt64(kKeyTime, mPrevSampleTimeUs);
364    buffer->meta_data()->setInt64(kKeyDriftTime, timeUs - mInitialReadTimeUs);
365    mPrevSampleTimeUs = timestampUs;
366    mNumFramesReceived += bufferSize / frameSize;
367    mBuffersReceived.push_back(buffer);
368    mFrameAvailableCondition.signal();
369}
370
371void AudioSource::trackMaxAmplitude(int16_t *data, int nSamples) {
372    for (int i = nSamples; i > 0; --i) {
373        int16_t value = *data++;
374        if (value < 0) {
375            value = -value;
376        }
377        if (mMaxAmplitude < value) {
378            mMaxAmplitude = value;
379        }
380    }
381}
382
383int16_t AudioSource::getMaxAmplitude() {
384    // First call activates the tracking.
385    if (!mTrackMaxAmplitude) {
386        mTrackMaxAmplitude = true;
387    }
388    int16_t value = mMaxAmplitude;
389    mMaxAmplitude = 0;
390    ALOGV("max amplitude since last call: %d", value);
391    return value;
392}
393
394}  // namespace android
395