diff options
Diffstat (limited to 'services/audioflinger/Threads.cpp')
| -rw-r--r-- | services/audioflinger/Threads.cpp | 59 |
1 files changed, 39 insertions, 20 deletions
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp index ee52fcb..0773534 100644 --- a/services/audioflinger/Threads.cpp +++ b/services/audioflinger/Threads.cpp @@ -139,7 +139,7 @@ static const int kPriorityFastMixer = 3; // FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or // N-buffering, so AudioFlinger could allocate the right amount of memory. // See the client's minBufCount and mNotificationFramesAct calculations for details. -static const int kFastTrackMultiplier = 2; +static const int kFastTrackMultiplier = 1; // ---------------------------------------------------------------------------- @@ -1327,7 +1327,7 @@ status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) // the track is newly added, make sure it fills up all its // buffers before playing. This is to ensure the client will // effectively get the latency it requested. - track->mFillingUpStatus = Track::FS_FILLING; + track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; track->mResetDone = false; track->mPresentationCompleteFrames = 0; mActiveTracks.add(track); @@ -2596,24 +2596,35 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac // app does not call stop() and relies on underrun to stop: // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed // during last round + size_t desiredFrames; + if (t->sampleRate() == mSampleRate) { + desiredFrames = mNormalFrameCount; + } else { + // +1 for rounding and +1 for additional sample needed for interpolation + desiredFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; + // add frames already consumed but not yet released by the resampler + // because cblk->framesReady() will include these frames + desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); + // the minimum track buffer size is normally twice the number of frames necessary + // to fill one buffer and the resampler should not leave more than one buffer worth + // of unreleased frames after each pass, but just in case... + ALOG_ASSERT(desiredFrames <= cblk->frameCount_); + } uint32_t minFrames = 1; if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { - if (t->sampleRate() == mSampleRate) { - minFrames = mNormalFrameCount; - } else { - // +1 for rounding and +1 for additional sample needed for interpolation - minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; - // add frames already consumed but not yet released by the resampler - // because cblk->framesReady() will include these frames - minFrames += mAudioMixer->getUnreleasedFrames(track->name()); - // the minimum track buffer size is normally twice the number of frames necessary - // to fill one buffer and the resampler should not leave more than one buffer worth - // of unreleased frames after each pass, but just in case... - ALOG_ASSERT(minFrames <= cblk->frameCount_); - } + minFrames = desiredFrames; } - if ((track->framesReady() >= minFrames) && track->isReady() && + // It's not safe to call framesReady() for a static buffer track, so assume it's ready + size_t framesReady; + if (track->sharedBuffer() == 0) { + framesReady = track->framesReady(); + } else if (track->isStopped()) { + framesReady = 0; + } else { + framesReady = 1; + } + if ((framesReady >= minFrames) && track->isReady() && !track->isPaused() && !track->isTerminated()) { ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, @@ -2664,7 +2675,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac // read original volumes with volume control float typeVolume = mStreamTypes[track->streamType()].volume; float v = masterVolume * typeVolume; - ServerProxy *proxy = track->mServerProxy; + AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; uint32_t vlr = proxy->getVolumeLR(); vl = vlr & 0xFFFF; vr = vlr >> 16; @@ -2737,7 +2748,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); // limit track sample rate to 2 x output sample rate, which changes at re-configuration uint32_t maxSampleRate = mSampleRate * 2; - uint32_t reqSampleRate = track->mServerProxy->getSampleRate(); + uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); if (reqSampleRate == 0) { reqSampleRate = mSampleRate; } else if (reqSampleRate > maxSampleRate) { @@ -2768,6 +2779,13 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac mixerStatus = MIXER_TRACKS_READY; } } else { + // only implemented for normal tracks, not fast tracks + if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { + // we missed desiredFrames whatever the actual number of frames missing was + cblk->u.mStreaming.mUnderrunFrames += desiredFrames; + // FIXME also wake futex so that underrun is noticed more quickly + (void) android_atomic_or(CBLK_UNDERRUN, &cblk->flags); + } // clear effect chain input buffer if an active track underruns to avoid sending // previous audio buffer again to effects chain = getEffectChain_l(track->sessionId()); @@ -3170,7 +3188,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep } else { float typeVolume = mStreamTypes[track->streamType()].volume; float v = mMasterVolume * typeVolume; - uint32_t vlr = track->mServerProxy->getVolumeLR(); + uint32_t vlr = track->mAudioTrackServerProxy->getVolumeLR(); float v_clamped = v * (vlr & 0xFFFF); if (v_clamped > MAX_GAIN) { v_clamped = MAX_GAIN; @@ -3696,7 +3714,8 @@ bool AudioFlinger::RecordThread::threadLoop() } buffer.frameCount = mFrameCount; - if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { + status_t status = mActiveTrack->getNextBuffer(&buffer); + if (CC_LIKELY(status == NO_ERROR)) { readOnce = true; size_t framesOut = buffer.frameCount; if (mResampler == NULL) { |