diff options
Diffstat (limited to 'media/libmedia/AudioTrack.cpp')
-rw-r--r-- | media/libmedia/AudioTrack.cpp | 59 |
1 files changed, 45 insertions, 14 deletions
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp index fe5cd9e..a9d6993 100644 --- a/media/libmedia/AudioTrack.cpp +++ b/media/libmedia/AudioTrack.cpp @@ -101,7 +101,8 @@ AudioTrack::AudioTrack( int notificationFrames, int sessionId, transfer_type transferType, - const audio_offload_info_t *offloadInfo) + const audio_offload_info_t *offloadInfo, + int uid) : mStatus(NO_INIT), mIsTimed(false), mPreviousPriority(ANDROID_PRIORITY_NORMAL), @@ -109,7 +110,8 @@ AudioTrack::AudioTrack( { mStatus = set(streamType, sampleRate, format, channelMask, frameCount, flags, cbf, user, notificationFrames, - 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo); + 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, + offloadInfo, uid); } AudioTrack::AudioTrack( @@ -124,7 +126,8 @@ AudioTrack::AudioTrack( int notificationFrames, int sessionId, transfer_type transferType, - const audio_offload_info_t *offloadInfo) + const audio_offload_info_t *offloadInfo, + int uid) : mStatus(NO_INIT), mIsTimed(false), mPreviousPriority(ANDROID_PRIORITY_NORMAL), @@ -132,7 +135,7 @@ AudioTrack::AudioTrack( { mStatus = set(streamType, sampleRate, format, channelMask, 0 /*frameCount*/, flags, cbf, user, notificationFrames, - sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo); + sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, uid); } AudioTrack::~AudioTrack() @@ -169,7 +172,8 @@ status_t AudioTrack::set( bool threadCanCallJava, int sessionId, transfer_type transferType, - const audio_offload_info_t *offloadInfo) + const audio_offload_info_t *offloadInfo, + int uid) { switch (transferType) { case TRANSFER_DEFAULT: @@ -313,6 +317,11 @@ status_t AudioTrack::set( mNotificationFramesReq = notificationFrames; mNotificationFramesAct = 0; mSessionId = sessionId; + if (uid == -1 || (IPCThreadState::self()->getCallingPid() != getpid())) { + mClientUid = IPCThreadState::self()->getCallingUid(); + } else { + mClientUid = uid; + } mAuxEffectId = 0; mFlags = flags; mCbf = cbf; @@ -594,6 +603,19 @@ uint32_t AudioTrack::getSampleRate() const } AutoMutex lock(mLock); + + // sample rate can be updated during playback by the offloaded decoder so we need to + // query the HAL and update if needed. +// FIXME use Proxy return channel to update the rate from server and avoid polling here + if (isOffloaded()) { + if (mOutput != 0) { + uint32_t sampleRate = 0; + status_t status = AudioSystem::getSamplingRate(mOutput, mStreamType, &sampleRate); + if (status == NO_ERROR) { + mSampleRate = sampleRate; + } + } + } return mSampleRate; } @@ -738,7 +760,7 @@ status_t AudioTrack::getPosition(uint32_t *position) const return NO_ERROR; } -status_t AudioTrack::getBufferPosition(size_t *position) +status_t AudioTrack::getBufferPosition(uint32_t *position) { if (mSharedBuffer == 0 || mIsTimed) { return INVALID_OPERATION; @@ -856,8 +878,15 @@ status_t AudioTrack::createTrack_l( } ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); + if ((flags & AUDIO_OUTPUT_FLAG_FAST) && sampleRate != afSampleRate) { + ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client due to mismatching sample rate (%d vs %d)", + sampleRate, afSampleRate); + flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); + } + // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where - // n = 1 fast track; nBuffering is ignored + // n = 1 fast track with single buffering; nBuffering is ignored + // n = 2 fast track with double buffering // n = 2 normal track, no sample rate conversion // n = 3 normal track, with sample rate conversion // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) @@ -886,7 +915,7 @@ status_t AudioTrack::createTrack_l( // More than 2 channels does not require stronger alignment than stereo alignment <<= 1; } - if (((size_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { + if (((uintptr_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { ALOGE("Invalid buffer alignment: address %p, channel count %u", sharedBuffer->pointer(), mChannelCount); return BAD_VALUE; @@ -962,6 +991,7 @@ status_t AudioTrack::createTrack_l( tid, &mSessionId, mName, + mClientUid, &status); if (track == 0) { @@ -996,9 +1026,11 @@ status_t AudioTrack::createTrack_l( ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); mAwaitBoost = true; if (sharedBuffer == 0) { - // double-buffering is not required for fast tracks, due to tighter scheduling - if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount) { - mNotificationFramesAct = frameCount; + // Theoretically double-buffering is not required for fast tracks, + // due to tighter scheduling. But in practice, to accommodate kernels with + // scheduling jitter, and apps with computation jitter, we use double-buffering. + if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { + mNotificationFramesAct = frameCount/nBuffering; } } } else { @@ -1212,7 +1244,7 @@ ssize_t AudioTrack::write(const void* buffer, size_t userSize) if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { // Sanity-check: user is most-likely passing an error code, and it would // make the return value ambiguous (actualSize vs error). - ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", buffer, userSize, userSize); + ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize); return BAD_VALUE; } @@ -1665,7 +1697,6 @@ status_t AudioTrack::restoreTrack_l(const char *from) // take the frames that will be lost by track recreation into account in saved position size_t position = mProxy->getPosition() + mProxy->getFramesFilled(); - mNewPosition = position + mUpdatePeriod; size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; result = createTrack_l(mStreamType, mSampleRate, @@ -1756,7 +1787,7 @@ status_t AudioTrack::dump(int fd, const Vector<String16>& args) const snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]); result.append(buffer); - snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, + 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); |