AudioSource.cpp revision 6e20bdf799a6f4efa6c42121a958634ea32ed5cc 
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    mStartTimeUs = 0;
88    int64_t startTimeUs;
89    if (params && params->findInt64(kKeyTime, &startTimeUs)) {
90        mStartTimeUs = startTimeUs;
91    }
92    status_t err = mRecord->start();
93
94    if (err == OK) {
95        mGroup = new MediaBufferGroup;
96        mGroup->add_buffer(new MediaBuffer(kMaxBufferSize));
97
98        mStarted = true;
99    }
100
101    return err;
102}
103
104status_t AudioSource::stop() {
105    if (!mStarted) {
106        return UNKNOWN_ERROR;
107    }
108
109    if (mInitCheck != OK) {
110        return NO_INIT;
111    }
112
113    mRecord->stop();
114
115    delete mGroup;
116    mGroup = NULL;
117
118    mStarted = false;
119
120    if (mCollectStats) {
121        LOGI("Total lost audio frames: %lld",
122            mTotalLostFrames + (mPrevLostBytes >> 1));
123    }
124
125    return OK;
126}
127
128sp<MetaData> AudioSource::getFormat() {
129    if (mInitCheck != OK) {
130        return 0;
131    }
132
133    sp<MetaData> meta = new MetaData;
134    meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_RAW);
135    meta->setInt32(kKeySampleRate, mRecord->getSampleRate());
136    meta->setInt32(kKeyChannelCount, mRecord->channelCount());
137    meta->setInt32(kKeyMaxInputSize, kMaxBufferSize);
138
139    return meta;
140}
141
142/*
143 * Returns -1 if frame skipping request is too long.
144 * Returns  0 if there is no need to skip frames.
145 * Returns  1 if we need to skip frames.
146 */
147static int skipFrame(int64_t timestampUs,
148        const MediaSource::ReadOptions *options) {
149
150    int64_t skipFrameUs;
151    if (!options || !options->getSkipFrame(&skipFrameUs)) {
152        return 0;
153    }
154
155    if (skipFrameUs <= timestampUs) {
156        return 0;
157    }
158
159    // Safe guard against the abuse of the kSkipFrame_Option.
160    if (skipFrameUs - timestampUs >= 1E6) {
161        LOGE("Frame skipping requested is way too long: %lld us",
162            skipFrameUs - timestampUs);
163
164        return -1;
165    }
166
167    LOGV("skipFrame: %lld us > timestamp: %lld us",
168        skipFrameUs, timestampUs);
169
170    return 1;
171
172}
173
174void AudioSource::rampVolume(
175        int32_t startFrame, int32_t rampDurationFrames,
176        uint8_t *data,   size_t bytes) {
177
178    const int32_t kShift = 14;
179    int32_t fixedMultiplier = (startFrame << kShift) / rampDurationFrames;
180    const int32_t nChannels = mRecord->channelCount();
181    int32_t stopFrame = startFrame + bytes / sizeof(int16_t);
182    int16_t *frame = (int16_t *) data;
183    if (stopFrame > rampDurationFrames) {
184        stopFrame = rampDurationFrames;
185    }
186
187    while (startFrame < stopFrame) {
188        if (nChannels == 1) {  // mono
189            frame[0] = (frame[0] * fixedMultiplier) >> kShift;
190            ++frame;
191            ++startFrame;
192        } else {               // stereo
193            frame[0] = (frame[0] * fixedMultiplier) >> kShift;
194            frame[1] = (frame[1] * fixedMultiplier) >> kShift;
195            frame += 2;
196            startFrame += 2;
197        }
198
199        // Update the multiplier every 4 frames
200        if ((startFrame & 3) == 0) {
201            fixedMultiplier = (startFrame << kShift) / rampDurationFrames;
202        }
203    }
204}
205
206status_t AudioSource::read(
207        MediaBuffer **out, const ReadOptions *options) {
208
209    if (mInitCheck != OK) {
210        return NO_INIT;
211    }
212
213    *out = NULL;
214
215    MediaBuffer *buffer;
216    CHECK_EQ(mGroup->acquire_buffer(&buffer), OK);
217
218    int err = 0;
219    while (mStarted) {
220
221        uint32_t numFramesRecorded;
222        mRecord->getPosition(&numFramesRecorded);
223
224
225        if (numFramesRecorded == 0 && mPrevSampleTimeUs == 0) {
226            // Initial delay
227            if (mStartTimeUs > 0) {
228                mStartTimeUs = systemTime() / 1000 - mStartTimeUs;
229            } else {
230                // Assume latency is constant.
231                mStartTimeUs += mRecord->latency() * 1000;
232            }
233            mPrevSampleTimeUs = mStartTimeUs;
234        }
235
236        uint32_t sampleRate = mRecord->getSampleRate();
237
238        // Insert null frames when lost frames are detected.
239        int64_t timestampUs = mPrevSampleTimeUs;
240        uint32_t numLostBytes = mRecord->getInputFramesLost() << 1;
241        numLostBytes += mPrevLostBytes;
242#if 0
243        // Simulate lost frames
244        numLostBytes = ((rand() * 1.0 / RAND_MAX)) * 2 * kMaxBufferSize;
245        numLostBytes &= 0xFFFFFFFE; // Alignment requirement
246
247        // Reduce the chance to lose
248        if (rand() * 1.0 / RAND_MAX >= 0.05) {
249            numLostBytes = 0;
250        }
251#endif
252        if (numLostBytes > 0) {
253            if (numLostBytes > kMaxBufferSize) {
254                mPrevLostBytes = numLostBytes - kMaxBufferSize;
255                numLostBytes = kMaxBufferSize;
256            }
257
258            CHECK_EQ(numLostBytes & 1, 0);
259            timestampUs += ((1000000LL * (numLostBytes >> 1)) +
260                    (sampleRate >> 1)) / sampleRate;
261
262            CHECK(timestampUs > mPrevSampleTimeUs);
263            if (mCollectStats) {
264                mTotalLostFrames += (numLostBytes >> 1);
265            }
266            if ((err = skipFrame(timestampUs, options)) == -1) {
267                buffer->release();
268                return UNKNOWN_ERROR;
269            } else if (err != 0) {
270                continue;
271            }
272            memset(buffer->data(), 0, numLostBytes);
273            buffer->set_range(0, numLostBytes);
274            buffer->meta_data()->setInt64(kKeyTime, mPrevSampleTimeUs);
275            mPrevSampleTimeUs = timestampUs;
276            *out = buffer;
277            return OK;
278        }
279
280        ssize_t n = mRecord->read(buffer->data(), buffer->size());
281        if (n < 0) {
282            buffer->release();
283            return (status_t)n;
284        }
285
286        int64_t recordDurationUs = (1000000LL * n >> 1) / sampleRate;
287        timestampUs += recordDurationUs;
288        if ((err = skipFrame(timestampUs, options)) == -1) {
289            buffer->release();
290            return UNKNOWN_ERROR;
291        } else if (err != 0) {
292            continue;
293        }
294
295        if (mPrevSampleTimeUs - mStartTimeUs < kAutoRampStartUs) {
296            // Mute the initial video recording signal
297            memset((uint8_t *) buffer->data(), 0, n);
298        } else if (mPrevSampleTimeUs - mStartTimeUs < kAutoRampStartUs + kAutoRampDurationUs) {
299            int32_t autoRampDurationFrames =
300                    (kAutoRampDurationUs * sampleRate + 500000LL) / 1000000LL;
301
302            int32_t autoRampStartFrames =
303                    (kAutoRampStartUs * sampleRate + 500000LL) / 1000000LL;
304
305            int32_t nFrames = numFramesRecorded - autoRampStartFrames;
306            rampVolume(nFrames, autoRampDurationFrames, (uint8_t *) buffer->data(), n);
307        }
308        if (mTrackMaxAmplitude) {
309            trackMaxAmplitude((int16_t *) buffer->data(), n >> 1);
310        }
311
312        buffer->meta_data()->setInt64(kKeyTime, mPrevSampleTimeUs);
313        CHECK(timestampUs > mPrevSampleTimeUs);
314        mPrevSampleTimeUs = timestampUs;
315        LOGV("initial delay: %lld, sample rate: %d, timestamp: %lld",
316                mStartTimeUs, sampleRate, timestampUs);
317
318        buffer->set_range(0, n);
319
320        *out = buffer;
321        return OK;
322    }
323
324    return OK;
325}
326
327void AudioSource::trackMaxAmplitude(int16_t *data, int nSamples) {
328    for (int i = nSamples; i > 0; --i) {
329        int16_t value = *data++;
330        if (value < 0) {
331            value = -value;
332        }
333        if (mMaxAmplitude < value) {
334            mMaxAmplitude = value;
335        }
336    }
337}
338
339int16_t AudioSource::getMaxAmplitude() {
340    // First call activates the tracking.
341    if (!mTrackMaxAmplitude) {
342        mTrackMaxAmplitude = true;
343    }
344    int16_t value = mMaxAmplitude;
345    mMaxAmplitude = 0;
346    LOGV("max amplitude since last call: %d", value);
347    return value;
348}
349
350}  // namespace android
351