diff options
Diffstat (limited to 'media/libmedia')
-rw-r--r-- | media/libmedia/AudioRecord.cpp | 10 | ||||
-rw-r--r-- | media/libmedia/AudioTrack.cpp | 139 | ||||
-rw-r--r-- | media/libmedia/AudioTrackShared.cpp | 10 | ||||
-rw-r--r-- | media/libmedia/ICrypto.cpp | 42 | ||||
-rw-r--r-- | media/libmedia/IMediaCodecList.cpp | 28 | ||||
-rw-r--r-- | media/libmedia/MediaCodecInfo.cpp | 11 |
6 files changed, 223 insertions, 17 deletions
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp index f4cdde2..5bbe786 100644 --- a/media/libmedia/AudioRecord.cpp +++ b/media/libmedia/AudioRecord.cpp @@ -596,15 +596,21 @@ release: return status; } -status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) +status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig) { if (audioBuffer == NULL) { + if (nonContig != NULL) { + *nonContig = 0; + } return BAD_VALUE; } if (mTransfer != TRANSFER_OBTAIN) { audioBuffer->frameCount = 0; audioBuffer->size = 0; audioBuffer->raw = NULL; + if (nonContig != NULL) { + *nonContig = 0; + } return INVALID_OPERATION; } @@ -623,7 +629,7 @@ status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) ALOGE("%s invalid waitCount %d", __func__, waitCount); requested = NULL; } - return obtainBuffer(audioBuffer, requested); + return obtainBuffer(audioBuffer, requested, NULL /*elapsed*/, nonContig); } status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp index 9e9ec5b..d32db7c 100644 --- a/media/libmedia/AudioTrack.cpp +++ b/media/libmedia/AudioTrack.cpp @@ -56,6 +56,24 @@ static int64_t getNowUs() return convertTimespecToUs(tv); } +// Must match similar computation in createTrack_l in Threads.cpp. +// TODO: Move to a common library +static size_t calculateMinFrameCount( + uint32_t afLatencyMs, uint32_t afFrameCount, uint32_t afSampleRate, + uint32_t sampleRate, float speed) +{ + // Ensure that buffer depth covers at least audio hardware latency + uint32_t minBufCount = afLatencyMs / ((1000 * afFrameCount) / afSampleRate); + if (minBufCount < 2) { + minBufCount = 2; + } + ALOGV("calculateMinFrameCount afLatency %u afFrameCount %u afSampleRate %u " + "sampleRate %u speed %f minBufCount: %u", + afLatencyMs, afFrameCount, afSampleRate, sampleRate, speed, minBufCount); + return minBufCount * sourceFramesNeededWithTimestretch( + sampleRate, afFrameCount, afSampleRate, speed); +} + // static status_t AudioTrack::getMinFrameCount( size_t* frameCount, @@ -94,13 +112,10 @@ status_t AudioTrack::getMinFrameCount( return status; } - // Ensure that buffer depth covers at least audio hardware latency - uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); - if (minBufCount < 2) { - minBufCount = 2; - } + // When called from createTrack, speed is 1.0f (normal speed). + // This is rechecked again on setting playback rate (TODO: on setting sample rate, too). + *frameCount = calculateMinFrameCount(afLatency, afFrameCount, afSampleRate, sampleRate, 1.0f); - *frameCount = minBufCount * sourceFramesNeeded(sampleRate, afFrameCount, afSampleRate); // The formula above should always produce a non-zero value under normal circumstances: // AudioTrack.SAMPLE_RATE_HZ_MIN <= sampleRate <= AudioTrack.SAMPLE_RATE_HZ_MAX. // Return error in the unlikely event that it does not, as that's part of the API contract. @@ -109,8 +124,8 @@ status_t AudioTrack::getMinFrameCount( streamType, sampleRate); return BAD_VALUE; } - ALOGV("getMinFrameCount=%zu: afFrameCount=%zu, minBufCount=%u, afSampleRate=%u, afLatency=%u", - *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); + ALOGV("getMinFrameCount=%zu: afFrameCount=%zu, afSampleRate=%u, afLatency=%u", + *frameCount, afFrameCount, afSampleRate, afLatency); return NO_ERROR; } @@ -360,6 +375,8 @@ status_t AudioTrack::set( return BAD_VALUE; } mSampleRate = sampleRate; + mSpeed = AUDIO_TIMESTRETCH_SPEED_NORMAL; + mPitch = AUDIO_TIMESTRETCH_PITCH_NORMAL; // Make copy of input parameter offloadInfo so that in the future: // (a) createTrack_l doesn't need it as an input parameter @@ -689,6 +706,7 @@ status_t AudioTrack::setSampleRate(uint32_t rate) if (rate == 0 || rate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { return BAD_VALUE; } + // TODO: Should we also check if the buffer size is compatible? mSampleRate = rate; mProxy->setSampleRate(rate); @@ -719,6 +737,42 @@ uint32_t AudioTrack::getSampleRate() const return mSampleRate; } +status_t AudioTrack::setPlaybackRate(float speed, float pitch) +{ + if (speed < AUDIO_TIMESTRETCH_SPEED_MIN + || speed > AUDIO_TIMESTRETCH_SPEED_MAX + || pitch < AUDIO_TIMESTRETCH_PITCH_MIN + || pitch > AUDIO_TIMESTRETCH_PITCH_MAX) { + return BAD_VALUE; + } + AutoMutex lock(mLock); + if (speed == mSpeed && pitch == mPitch) { + return NO_ERROR; + } + if (mIsTimed || isOffloadedOrDirect_l()) { + return INVALID_OPERATION; + } + if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { + return INVALID_OPERATION; + } + // Check if the buffer size is compatible. + if (!isSampleRateSpeedAllowed_l(mSampleRate, speed)) { + ALOGV("setPlaybackRate(%f, %f) failed", speed, pitch); + return BAD_VALUE; + } + mSpeed = speed; + mPitch = pitch; + mProxy->setPlaybackRate(speed, pitch); + return NO_ERROR; +} + +void AudioTrack::getPlaybackRate(float *speed, float *pitch) const +{ + AutoMutex lock(mLock); + *speed = mSpeed; + *pitch = mPitch; +} + status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) { if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { @@ -1086,8 +1140,16 @@ status_t AudioTrack::createTrack_l() // there _is_ a frameCount parameter. We silently ignore it. frameCount = mSharedBuffer->size() / mFrameSize; } else { - // For fast and normal streaming tracks, - // the frame count calculations and checks are done by server + // For fast tracks the frame count calculations and checks are done by server + + if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) == 0) { + // for normal tracks precompute the frame count based on speed. + const size_t minFrameCount = calculateMinFrameCount( + afLatency, afFrameCount, afSampleRate, mSampleRate, mSpeed); + if (frameCount < minFrameCount) { + frameCount = minFrameCount; + } + } } IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; @@ -1230,6 +1292,7 @@ status_t AudioTrack::createTrack_l() } mAudioTrack->attachAuxEffect(mAuxEffectId); + // FIXME doesn't take into account speed or future sample rate changes (until restoreTrack) // FIXME don't believe this lie mLatency = afLatency + (1000*frameCount) / mSampleRate; @@ -1255,6 +1318,7 @@ status_t AudioTrack::createTrack_l() mProxy->setSendLevel(mSendLevel); mProxy->setSampleRate(mSampleRate); + mProxy->setPlaybackRate(mSpeed, mPitch); mProxy->setMinimum(mNotificationFramesAct); mDeathNotifier = new DeathNotifier(this); @@ -1274,12 +1338,18 @@ release: status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig) { if (audioBuffer == NULL) { + if (nonContig != NULL) { + *nonContig = 0; + } return BAD_VALUE; } if (mTransfer != TRANSFER_OBTAIN) { audioBuffer->frameCount = 0; audioBuffer->size = 0; audioBuffer->raw = NULL; + if (nonContig != NULL) { + *nonContig = 0; + } return INVALID_OPERATION; } @@ -1617,6 +1687,7 @@ nsecs_t AudioTrack::processAudioBuffer() // Cache other fields that will be needed soon uint32_t sampleRate = mSampleRate; + float speed = mSpeed; uint32_t notificationFrames = mNotificationFramesAct; if (mRefreshRemaining) { mRefreshRemaining = false; @@ -1745,7 +1816,7 @@ nsecs_t AudioTrack::processAudioBuffer() if (minFrames != (uint32_t) ~0) { // This "fudge factor" avoids soaking CPU, and compensates for late progress by server static const nsecs_t kFudgeNs = 10000000LL; // 10 ms - ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; + ns = ((double)minFrames * 1000000000) / ((double)sampleRate * speed) + kFudgeNs; } // If not supplying data by EVENT_MORE_DATA, then we're done @@ -1786,7 +1857,8 @@ nsecs_t AudioTrack::processAudioBuffer() if (mRetryOnPartialBuffer && !isOffloaded()) { mRetryOnPartialBuffer = false; if (avail < mRemainingFrames) { - int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; + int64_t myns = ((double)(mRemainingFrames - avail) * 1100000000) + / ((double)sampleRate * speed); if (ns < 0 || myns < ns) { ns = myns; } @@ -1841,7 +1913,7 @@ nsecs_t AudioTrack::processAudioBuffer() // that total to a sum == notificationFrames. if (0 < misalignment && misalignment <= mRemainingFrames) { mRemainingFrames = misalignment; - return (mRemainingFrames * 1100000000LL) / sampleRate; + return ((double)mRemainingFrames * 1100000000) / ((double)sampleRate * speed); } #endif @@ -1936,6 +2008,41 @@ uint32_t AudioTrack::updateAndGetPosition_l() return mPosition += (uint32_t) delta; } +bool AudioTrack::isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed) const +{ + // applicable for mixing tracks only (not offloaded or direct) + if (mStaticProxy != 0) { + return true; // static tracks do not have issues with buffer sizing. + } + status_t status; + uint32_t afLatency; + status = AudioSystem::getLatency(mOutput, &afLatency); + if (status != NO_ERROR) { + ALOGE("getLatency(%d) failed status %d", mOutput, status); + return false; + } + + size_t afFrameCount; + status = AudioSystem::getFrameCount(mOutput, &afFrameCount); + if (status != NO_ERROR) { + ALOGE("getFrameCount(output=%d) status %d", mOutput, status); + return false; + } + + uint32_t afSampleRate; + status = AudioSystem::getSamplingRate(mOutput, &afSampleRate); + if (status != NO_ERROR) { + ALOGE("getSamplingRate(output=%d) status %d", mOutput, status); + return false; + } + + const size_t minFrameCount = + calculateMinFrameCount(afLatency, afFrameCount, afSampleRate, sampleRate, speed); + ALOGV("isSampleRateSpeedAllowed_l mFrameCount %zu minFrameCount %zu", + mFrameCount, minFrameCount); + return mFrameCount >= minFrameCount; +} + status_t AudioTrack::setParameters(const String8& keyValuePairs) { AutoMutex lock(mLock); @@ -2001,7 +2108,8 @@ status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) return WOULD_BLOCK; // stale timestamp time, occurs before start. } const int64_t deltaTimeUs = timestampTimeUs - mStartUs; - const int64_t deltaPositionByUs = timestamp.mPosition * 1000000LL / mSampleRate; + const int64_t deltaPositionByUs = (double)timestamp.mPosition * 1000000 + / ((double)mSampleRate * mSpeed); if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) { // Verify that the counter can't count faster than the sample rate @@ -2088,7 +2196,8 @@ status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%zu)\n", mFormat, mChannelCount, mFrameCount); result.append(buffer); - snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); + snprintf(buffer, 255, " sample rate(%u), speed(%f), status(%d)\n", + mSampleRate, mSpeed, mStatus); result.append(buffer); snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); result.append(buffer); diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp index 6d5f1af..ba67b40 100644 --- a/media/libmedia/AudioTrackShared.cpp +++ b/media/libmedia/AudioTrackShared.cpp @@ -793,6 +793,16 @@ void AudioTrackServerProxy::tallyUnderrunFrames(uint32_t frameCount) (void) android_atomic_or(CBLK_UNDERRUN, &cblk->mFlags); } +void AudioTrackServerProxy::getPlaybackRate(float *speed, float *pitch) +{ // do not call from multiple threads without holding lock + AudioTrackPlaybackRate playbackRate; + if (mPlaybackRateObserver.poll(playbackRate)) { + mPlaybackRate = playbackRate; + } + *speed = mPlaybackRate.mSpeed; + *pitch = mPlaybackRate.mPitch; +} + // --------------------------------------------------------------------------- StaticAudioTrackServerProxy::StaticAudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers, diff --git a/media/libmedia/ICrypto.cpp b/media/libmedia/ICrypto.cpp index 23308c1..9246a7c 100644 --- a/media/libmedia/ICrypto.cpp +++ b/media/libmedia/ICrypto.cpp @@ -35,6 +35,7 @@ enum { REQUIRES_SECURE_COMPONENT, DECRYPT, NOTIFY_RESOLUTION, + SET_MEDIADRM_SESSION, }; struct BpCrypto : public BpInterface<ICrypto> { @@ -161,7 +162,28 @@ struct BpCrypto : public BpInterface<ICrypto> { remote()->transact(NOTIFY_RESOLUTION, data, &reply); } + virtual status_t setMediaDrmSession(const Vector<uint8_t> &sessionId) { + Parcel data, reply; + data.writeInterfaceToken(ICrypto::getInterfaceDescriptor()); + + writeVector(data, sessionId); + remote()->transact(SET_MEDIADRM_SESSION, data, &reply); + + return reply.readInt32(); + } + private: + void readVector(Parcel &reply, Vector<uint8_t> &vector) const { + uint32_t size = reply.readInt32(); + vector.insertAt((size_t)0, size); + reply.read(vector.editArray(), size); + } + + void writeVector(Parcel &data, Vector<uint8_t> const &vector) const { + data.writeInt32(vector.size()); + data.write(vector.array(), vector.size()); + } + DISALLOW_EVIL_CONSTRUCTORS(BpCrypto); }; @@ -169,6 +191,17 @@ IMPLEMENT_META_INTERFACE(Crypto, "android.hardware.ICrypto"); //////////////////////////////////////////////////////////////////////////////// +void BnCrypto::readVector(const Parcel &data, Vector<uint8_t> &vector) const { + uint32_t size = data.readInt32(); + vector.insertAt((size_t)0, size); + data.read(vector.editArray(), size); +} + +void BnCrypto::writeVector(Parcel *reply, Vector<uint8_t> const &vector) const { + reply->writeInt32(vector.size()); + reply->write(vector.array(), vector.size()); +} + status_t BnCrypto::onTransact( uint32_t code, const Parcel &data, Parcel *reply, uint32_t flags) { switch (code) { @@ -317,6 +350,15 @@ status_t BnCrypto::onTransact( return OK; } + case SET_MEDIADRM_SESSION: + { + CHECK_INTERFACE(IDrm, data, reply); + Vector<uint8_t> sessionId; + readVector(data, sessionId); + reply->writeInt32(setMediaDrmSession(sessionId)); + return OK; + } + default: return BBinder::onTransact(code, data, reply, flags); } diff --git a/media/libmedia/IMediaCodecList.cpp b/media/libmedia/IMediaCodecList.cpp index 80020db..e2df104 100644 --- a/media/libmedia/IMediaCodecList.cpp +++ b/media/libmedia/IMediaCodecList.cpp @@ -30,6 +30,7 @@ enum { CREATE = IBinder::FIRST_CALL_TRANSACTION, COUNT_CODECS, GET_CODEC_INFO, + GET_GLOBAL_SETTINGS, FIND_CODEC_BY_TYPE, FIND_CODEC_BY_NAME, }; @@ -64,6 +65,19 @@ public: } } + virtual const sp<AMessage> getGlobalSettings() const + { + Parcel data, reply; + data.writeInterfaceToken(IMediaCodecList::getInterfaceDescriptor()); + remote()->transact(GET_GLOBAL_SETTINGS, data, &reply); + status_t err = reply.readInt32(); + if (err == OK) { + return AMessage::FromParcel(reply); + } else { + return NULL; + } + } + virtual ssize_t findCodecByType( const char *type, bool encoder, size_t startIndex = 0) const { @@ -125,6 +139,20 @@ status_t BnMediaCodecList::onTransact( } break; + case GET_GLOBAL_SETTINGS: + { + CHECK_INTERFACE(IMediaCodecList, data, reply); + const sp<AMessage> info = getGlobalSettings(); + if (info != NULL) { + reply->writeInt32(OK); + info->writeToParcel(reply); + } else { + reply->writeInt32(-ERANGE); + } + return NO_ERROR; + } + break; + case FIND_CODEC_BY_TYPE: { CHECK_INTERFACE(IMediaCodecList, data, reply); diff --git a/media/libmedia/MediaCodecInfo.cpp b/media/libmedia/MediaCodecInfo.cpp index 7b4c4e2..8d3fa7b 100644 --- a/media/libmedia/MediaCodecInfo.cpp +++ b/media/libmedia/MediaCodecInfo.cpp @@ -206,6 +206,17 @@ status_t MediaCodecInfo::addMime(const char *mime) { return OK; } +status_t MediaCodecInfo::updateMime(const char *mime) { + ssize_t ix = getCapabilityIndex(mime); + if (ix < 0) { + ALOGE("updateMime mime not found %s", mime); + return -EINVAL; + } + + mCurrentCaps = mCaps.valueAt(ix); + return OK; +} + void MediaCodecInfo::removeMime(const char *mime) { ssize_t ix = getCapabilityIndex(mime); if (ix >= 0) { |