AudioSource.cpp revision 3856b090cd04ba5dd4a59a12430ed724d5995909 
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            LOGW("AudioRecord reported overrun!");
41            break;
42        }
43        default:
44            // does nothing
45            break;
46    }
47}
48
49AudioSource::AudioSource(
50        int 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    uint32_t flags = AudioRecord::RECORD_AGC_ENABLE |
60                     AudioRecord::RECORD_NS_ENABLE  |
61                     AudioRecord::RECORD_IIR_ENABLE;
62    mRecord = new AudioRecord(
63                inputSource, sampleRate, AUDIO_FORMAT_PCM_16_BIT,
64                channels > 1? AUDIO_CHANNEL_IN_STEREO: AUDIO_CHANNEL_IN_MONO,
65                4 * kMaxBufferSize / sizeof(int16_t), /* Enable ping-pong buffers */
66                flags,
67                AudioRecordCallbackFunction,
68                this);
69
70    mInitCheck = mRecord->initCheck();
71}
72
73AudioSource::~AudioSource() {
74    if (mStarted) {
75        stop();
76    }
77
78    delete mRecord;
79    mRecord = NULL;
80}
81
82status_t AudioSource::initCheck() const {
83    return mInitCheck;
84}
85
86status_t AudioSource::start(MetaData *params) {
87    Mutex::Autolock autoLock(mLock);
88    if (mStarted) {
89        return UNKNOWN_ERROR;
90    }
91
92    if (mInitCheck != OK) {
93        return NO_INIT;
94    }
95
96    mTrackMaxAmplitude = false;
97    mMaxAmplitude = 0;
98    mInitialReadTimeUs = 0;
99    mStartTimeUs = 0;
100    int64_t startTimeUs;
101    if (params && params->findInt64(kKeyTime, &startTimeUs)) {
102        mStartTimeUs = startTimeUs;
103    }
104    status_t err = mRecord->start();
105    if (err == OK) {
106        mStarted = true;
107    } else {
108        delete mRecord;
109        mRecord = NULL;
110    }
111
112
113    return err;
114}
115
116void AudioSource::releaseQueuedFrames_l() {
117    ALOGV("releaseQueuedFrames_l");
118    List<MediaBuffer *>::iterator it;
119    while (!mBuffersReceived.empty()) {
120        it = mBuffersReceived.begin();
121        (*it)->release();
122        mBuffersReceived.erase(it);
123    }
124}
125
126void AudioSource::waitOutstandingEncodingFrames_l() {
127    ALOGV("waitOutstandingEncodingFrames_l: %lld", mNumClientOwnedBuffers);
128    while (mNumClientOwnedBuffers > 0) {
129        mFrameEncodingCompletionCondition.wait(mLock);
130    }
131}
132
133status_t AudioSource::stop() {
134    Mutex::Autolock autoLock(mLock);
135    if (!mStarted) {
136        return UNKNOWN_ERROR;
137    }
138
139    if (mInitCheck != OK) {
140        return NO_INIT;
141    }
142
143    mStarted = false;
144    mRecord->stop();
145    waitOutstandingEncodingFrames_l();
146    releaseQueuedFrames_l();
147
148    return OK;
149}
150
151sp<MetaData> AudioSource::getFormat() {
152    Mutex::Autolock autoLock(mLock);
153    if (mInitCheck != OK) {
154        return 0;
155    }
156
157    sp<MetaData> meta = new MetaData;
158    meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_RAW);
159    meta->setInt32(kKeySampleRate, mSampleRate);
160    meta->setInt32(kKeyChannelCount, mRecord->channelCount());
161    meta->setInt32(kKeyMaxInputSize, kMaxBufferSize);
162
163    return meta;
164}
165
166void AudioSource::rampVolume(
167        int32_t startFrame, int32_t rampDurationFrames,
168        uint8_t *data,   size_t bytes) {
169
170    const int32_t kShift = 14;
171    int32_t fixedMultiplier = (startFrame << kShift) / rampDurationFrames;
172    const int32_t nChannels = mRecord->channelCount();
173    int32_t stopFrame = startFrame + bytes / sizeof(int16_t);
174    int16_t *frame = (int16_t *) data;
175    if (stopFrame > rampDurationFrames) {
176        stopFrame = rampDurationFrames;
177    }
178
179    while (startFrame < stopFrame) {
180        if (nChannels == 1) {  // mono
181            frame[0] = (frame[0] * fixedMultiplier) >> kShift;
182            ++frame;
183            ++startFrame;
184        } else {               // stereo
185            frame[0] = (frame[0] * fixedMultiplier) >> kShift;
186            frame[1] = (frame[1] * fixedMultiplier) >> kShift;
187            frame += 2;
188            startFrame += 2;
189        }
190
191        // Update the multiplier every 4 frames
192        if ((startFrame & 3) == 0) {
193            fixedMultiplier = (startFrame << kShift) / rampDurationFrames;
194        }
195    }
196}
197
198status_t AudioSource::read(
199        MediaBuffer **out, const ReadOptions *options) {
200    Mutex::Autolock autoLock(mLock);
201    *out = NULL;
202
203    if (mInitCheck != OK) {
204        return NO_INIT;
205    }
206
207    while (mStarted && mBuffersReceived.empty()) {
208        mFrameAvailableCondition.wait(mLock);
209    }
210    if (!mStarted) {
211        return OK;
212    }
213    MediaBuffer *buffer = *mBuffersReceived.begin();
214    mBuffersReceived.erase(mBuffersReceived.begin());
215    ++mNumClientOwnedBuffers;
216    buffer->setObserver(this);
217    buffer->add_ref();
218
219    // Mute/suppress the recording sound
220    int64_t timeUs;
221    CHECK(buffer->meta_data()->findInt64(kKeyTime, &timeUs));
222    int64_t elapsedTimeUs = timeUs - mStartTimeUs;
223    if (elapsedTimeUs < kAutoRampStartUs) {
224        memset((uint8_t *) buffer->data(), 0, buffer->range_length());
225    } else if (elapsedTimeUs < kAutoRampStartUs + kAutoRampDurationUs) {
226        int32_t autoRampDurationFrames =
227                    (kAutoRampDurationUs * mSampleRate + 500000LL) / 1000000LL;
228
229        int32_t autoRampStartFrames =
230                    (kAutoRampStartUs * mSampleRate + 500000LL) / 1000000LL;
231
232        int32_t nFrames = mNumFramesReceived - autoRampStartFrames;
233        rampVolume(nFrames, autoRampDurationFrames,
234                (uint8_t *) buffer->data(), buffer->range_length());
235    }
236
237    // Track the max recording signal amplitude.
238    if (mTrackMaxAmplitude) {
239        trackMaxAmplitude(
240            (int16_t *) buffer->data(), buffer->range_length() >> 1);
241    }
242
243    *out = buffer;
244    return OK;
245}
246
247void AudioSource::signalBufferReturned(MediaBuffer *buffer) {
248    ALOGV("signalBufferReturned: %p", buffer->data());
249    Mutex::Autolock autoLock(mLock);
250    --mNumClientOwnedBuffers;
251    buffer->setObserver(0);
252    buffer->release();
253    mFrameEncodingCompletionCondition.signal();
254    return;
255}
256
257status_t AudioSource::dataCallbackTimestamp(
258        const AudioRecord::Buffer& audioBuffer, int64_t timeUs) {
259    ALOGV("dataCallbackTimestamp: %lld us", timeUs);
260    Mutex::Autolock autoLock(mLock);
261    if (!mStarted) {
262        LOGW("Spurious callback from AudioRecord. Drop the audio data.");
263        return OK;
264    }
265
266    // Drop retrieved and previously lost audio data.
267    if (mNumFramesReceived == 0 && timeUs < mStartTimeUs) {
268        mRecord->getInputFramesLost();
269        ALOGV("Drop audio data at %lld/%lld us", timeUs, mStartTimeUs);
270        return OK;
271    }
272
273    if (mNumFramesReceived == 0 && mPrevSampleTimeUs == 0) {
274        mInitialReadTimeUs = timeUs;
275        // Initial delay
276        if (mStartTimeUs > 0) {
277            mStartTimeUs = timeUs - mStartTimeUs;
278        } else {
279            // Assume latency is constant.
280            mStartTimeUs += mRecord->latency() * 1000;
281        }
282        mPrevSampleTimeUs = mStartTimeUs;
283    }
284
285    int64_t timestampUs = mPrevSampleTimeUs;
286
287    size_t numLostBytes = 0;
288    if (mNumFramesReceived > 0) {  // Ignore earlier frame lost
289        // getInputFramesLost() returns the number of lost frames.
290        // Convert number of frames lost to number of bytes lost.
291        numLostBytes = mRecord->getInputFramesLost() * mRecord->frameSize();
292    }
293
294    CHECK_EQ(numLostBytes & 1, 0u);
295    CHECK_EQ(audioBuffer.size & 1, 0u);
296    size_t bufferSize = numLostBytes + audioBuffer.size;
297    MediaBuffer *buffer = new MediaBuffer(bufferSize);
298    if (numLostBytes > 0) {
299        memset(buffer->data(), 0, numLostBytes);
300        memcpy((uint8_t *) buffer->data() + numLostBytes,
301                    audioBuffer.i16, audioBuffer.size);
302    } else {
303        if (audioBuffer.size == 0) {
304            LOGW("Nothing is available from AudioRecord callback buffer");
305            buffer->release();
306            return OK;
307        }
308        memcpy((uint8_t *) buffer->data(),
309                audioBuffer.i16, audioBuffer.size);
310    }
311
312    buffer->set_range(0, bufferSize);
313    timestampUs += ((1000000LL * (bufferSize >> 1)) +
314                    (mSampleRate >> 1)) / mSampleRate;
315
316    if (mNumFramesReceived == 0) {
317        buffer->meta_data()->setInt64(kKeyAnchorTime, mStartTimeUs);
318    }
319    buffer->meta_data()->setInt64(kKeyTime, mPrevSampleTimeUs);
320    buffer->meta_data()->setInt64(kKeyDriftTime, timeUs - mInitialReadTimeUs);
321    mPrevSampleTimeUs = timestampUs;
322    mNumFramesReceived += buffer->range_length() / sizeof(int16_t);
323    mBuffersReceived.push_back(buffer);
324    mFrameAvailableCondition.signal();
325
326    return OK;
327}
328
329void AudioSource::trackMaxAmplitude(int16_t *data, int nSamples) {
330    for (int i = nSamples; i > 0; --i) {
331        int16_t value = *data++;
332        if (value < 0) {
333            value = -value;
334        }
335        if (mMaxAmplitude < value) {
336            mMaxAmplitude = value;
337        }
338    }
339}
340
341int16_t AudioSource::getMaxAmplitude() {
342    // First call activates the tracking.
343    if (!mTrackMaxAmplitude) {
344        mTrackMaxAmplitude = true;
345    }
346    int16_t value = mMaxAmplitude;
347    mMaxAmplitude = 0;
348    ALOGV("max amplitude since last call: %d", value);
349    return value;
350}
351
352}  // namespace android
353