diff options
| -rw-r--r-- | services/audioflinger/AudioFlinger.cpp | 147 | ||||
| -rw-r--r-- | services/audioflinger/AudioFlinger.h | 9 | ||||
| -rw-r--r-- | services/audioflinger/AudioPolicyService.cpp | 1 |
3 files changed, 33 insertions, 124 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp index 0ebea76..2cfc3e8 100644 --- a/services/audioflinger/AudioFlinger.cpp +++ b/services/audioflinger/AudioFlinger.cpp @@ -2013,11 +2013,9 @@ void AudioFlinger::PlaybackThread::readOutputParameters() mNormalFrameCount = (mNormalFrameCount + 15) & ~15; ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); - // FIXME - Current mixer implementation only supports stereo output: Always - // Allocate a stereo buffer even if HW output is mono. delete[] mMixBuffer; - mMixBuffer = new int16_t[mNormalFrameCount * 2]; - memset(mMixBuffer, 0, mNormalFrameCount * 2 * sizeof(int16_t)); + mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; + memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); // force reconfiguration of effect chains and engines to take new buffer size and audio // parameters into account @@ -2675,21 +2673,31 @@ void AudioFlinger::PlaybackThread::threadLoop_write() // FIXME rewrite to reduce number of system calls mLastWriteTime = systemTime(); mInWrite = true; + int bytesWritten; + // If an NBAIO sink is present, use it to write the normal mixer's submix + if (mNormalSink != 0) { #define mBitShift 2 // FIXME - size_t count = mixBufferSize >> mBitShift; + size_t count = mixBufferSize >> mBitShift; #if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) - Tracer::traceBegin(ATRACE_TAG, "write"); + Tracer::traceBegin(ATRACE_TAG, "write"); #endif - ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); + ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); #if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) - Tracer::traceEnd(ATRACE_TAG); + Tracer::traceEnd(ATRACE_TAG); #endif - if (framesWritten > 0) { - size_t bytesWritten = framesWritten << mBitShift; - mBytesWritten += bytesWritten; + if (framesWritten > 0) { + bytesWritten = framesWritten << mBitShift; + } else { + bytesWritten = framesWritten; + } + // otherwise use the HAL / AudioStreamOut directly + } else { + // Direct output thread. + bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); } + if (bytesWritten > 0) mBytesWritten += mixBufferSize; mNumWrites++; mInWrite = false; } @@ -3570,7 +3578,6 @@ AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& aud AudioStreamOut* output, audio_io_handle_t id, uint32_t device) : PlaybackThread(audioFlinger, output, id, device, DIRECT) // mLeftVolFloat, mRightVolFloat - // mLeftVolShort, mRightVolShort { } @@ -3597,7 +3604,13 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep // The first time a track is added we wait // for all its buffers to be filled before processing it - if (cblk->framesReady() && track->isReady() && + uint32_t minFrames; + if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { + minFrames = mNormalFrameCount; + } else { + minFrames = 1; + } + if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && !track->isTerminated()) { //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); @@ -3605,16 +3618,11 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep if (track->mFillingUpStatus == Track::FS_FILLED) { track->mFillingUpStatus = Track::FS_ACTIVE; mLeftVolFloat = mRightVolFloat = 0; - mLeftVolShort = mRightVolShort = 0; if (track->mState == TrackBase::RESUMING) { track->mState = TrackBase::ACTIVE; - rampVolume = true; } - } else if (cblk->server != 0) { - // If the track is stopped before the first frame was mixed, - // do not apply ramp - rampVolume = true; } + // compute volume for this track float left, right; if (track->isMuted() || mMasterMute || track->isPausing() || @@ -3639,13 +3647,6 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep mLeftVolFloat = left; mRightVolFloat = right; - // If audio HAL implements volume control, - // force software volume to nominal value - if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { - left = 1.0f; - right = 1.0f; - } - // Convert volumes from float to 8.24 uint32_t vl = (uint32_t)(left * (1 << 24)); uint32_t vr = (uint32_t)(right * (1 << 24)); @@ -3655,22 +3656,11 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep // there is one, the track is connected to it if (!mEffectChains.isEmpty()) { // Do not ramp volume if volume is controlled by effect - if (mEffectChains[0]->setVolume_l(&vl, &vr)) { - rampVolume = false; - } + mEffectChains[0]->setVolume_l(&vl, &vr); + left = (float)vl / (1 << 24); + right = (float)vr / (1 << 24); } - - // Convert volumes from 8.24 to 4.12 format - uint32_t v_clamped = (vl + (1 << 11)) >> 12; - if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; - leftVol = (uint16_t)v_clamped; - v_clamped = (vr + (1 << 11)) >> 12; - if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; - rightVol = (uint16_t)v_clamped; - } else { - leftVol = mLeftVolShort; - rightVol = mRightVolShort; - rampVolume = false; + mOutput->stream->set_volume(mOutput->stream, left, right); } // reset retry count @@ -3685,7 +3675,8 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep } //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); - if (track->isTerminated() || track->isStopped() || track->isPaused()) { + if ((track->sharedBuffer() != 0) || track->isTerminated() || + track->isStopped() || track->isPaused()) { // We have consumed all the buffers of this track. // Remove it from the list of active tracks. // TODO: implement behavior for compressed audio @@ -3752,78 +3743,6 @@ void AudioFlinger::DirectOutputThread::threadLoop_mix() standbyTime = systemTime() + standbyDelay; mActiveTrack.clear(); - // apply volume - - // Do not apply volume on compressed audio - if (!audio_is_linear_pcm(mFormat)) { - return; - } - - // convert to signed 16 bit before volume calculation - if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { - size_t count = mFrameCount * mChannelCount; - uint8_t *src = (uint8_t *)mMixBuffer + count-1; - int16_t *dst = mMixBuffer + count-1; - while (count--) { - *dst-- = (int16_t)(*src--^0x80) << 8; - } - } - - frameCount = mFrameCount; - int16_t *out = mMixBuffer; - if (rampVolume) { - if (mChannelCount == 1) { - int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; - int32_t vlInc = d / (int32_t)frameCount; - int32_t vl = ((int32_t)mLeftVolShort << 16); - do { - out[0] = clamp16(mul(out[0], vl >> 16) >> 12); - out++; - vl += vlInc; - } while (--frameCount); - - } else { - int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; - int32_t vlInc = d / (int32_t)frameCount; - d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; - int32_t vrInc = d / (int32_t)frameCount; - int32_t vl = ((int32_t)mLeftVolShort << 16); - int32_t vr = ((int32_t)mRightVolShort << 16); - do { - out[0] = clamp16(mul(out[0], vl >> 16) >> 12); - out[1] = clamp16(mul(out[1], vr >> 16) >> 12); - out += 2; - vl += vlInc; - vr += vrInc; - } while (--frameCount); - } - } else { - if (mChannelCount == 1) { - do { - out[0] = clamp16(mul(out[0], leftVol) >> 12); - out++; - } while (--frameCount); - } else { - do { - out[0] = clamp16(mul(out[0], leftVol) >> 12); - out[1] = clamp16(mul(out[1], rightVol) >> 12); - out += 2; - } while (--frameCount); - } - } - - // convert back to unsigned 8 bit after volume calculation - if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { - size_t count = mFrameCount * mChannelCount; - int16_t *src = mMixBuffer; - uint8_t *dst = (uint8_t *)mMixBuffer; - while (count--) { - *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; - } - } - - mLeftVolShort = leftVol; - mRightVolShort = rightVol; } void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h index 6da5802..a0e0ea5 100644 --- a/services/audioflinger/AudioFlinger.h +++ b/services/audioflinger/AudioFlinger.h @@ -1211,17 +1211,8 @@ public: virtual void threadLoop_sleepTime(); // volumes last sent to audio HAL with stream->set_volume() - // FIXME use standard representation and names float mLeftVolFloat; float mRightVolFloat; - uint16_t mLeftVolShort; - uint16_t mRightVolShort; - - // FIXME rename these former local variables of threadLoop to standard names - // next 3 were local to the while !exitingPending loop - bool rampVolume; - uint16_t leftVol; - uint16_t rightVol; private: // prepareTracks_l() tells threadLoop_mix() the name of the single active track diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp index ca25ba9..0d13970 100644 --- a/services/audioflinger/AudioPolicyService.cpp +++ b/services/audioflinger/AudioPolicyService.cpp @@ -1375,7 +1375,6 @@ static audio_io_handle_t aps_open_output_on_module(void *service, ALOGW("%s: could not get AudioFlinger", __func__); return 0; } - ALOGW("%s: %d", __func__, module); return af->openOutput(module, pDevices, pSamplingRate, pFormat, pChannelMask, pLatencyMs, flags); } |