Code drop from //branches/cupcake/...@124589
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 298170a..d4f2e5a 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -2,16 +2,16 @@
**
** Copyright 2007, The Android Open Source Project
**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
+** Licensed under the Apache License, Version 2.0 (the "License");
+** you may not use this file except in compliance with the License.
+** You may obtain a copy of the License at
**
-** http://www.apache.org/licenses/LICENSE-2.0
+** http://www.apache.org/licenses/LICENSE-2.0
**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
+** Unless required by applicable law or agreed to in writing, software
+** distributed under the License is distributed on an "AS IS" BASIS,
+** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+** See the License for the specific language governing permissions and
** limitations under the License.
*/
@@ -21,6 +21,7 @@
#include <stdint.h>
#include <sys/types.h>
+#include <limits.h>
#include <sched.h>
#include <sys/resource.h>
@@ -44,19 +45,6 @@
// ---------------------------------------------------------------------------
-static volatile size_t gFrameCount = 0;
-
-size_t AudioTrack::frameCount()
-{
- if (gFrameCount) return gFrameCount;
- const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
- if (af == 0) return PERMISSION_DENIED;
- gFrameCount = af->frameCount();
- return gFrameCount;
-}
-
-// ---------------------------------------------------------------------------
-
AudioTrack::AudioTrack()
: mStatus(NO_INIT)
{
@@ -67,24 +55,44 @@
uint32_t sampleRate,
int format,
int channelCount,
- int bufferCount,
+ int frameCount,
uint32_t flags,
- callback_t cbf, void* user)
+ callback_t cbf,
+ void* user,
+ int notificationFrames)
: mStatus(NO_INIT)
{
mStatus = set(streamType, sampleRate, format, channelCount,
- bufferCount, flags, cbf, user);
+ frameCount, flags, cbf, user, notificationFrames, 0);
+}
+
+AudioTrack::AudioTrack(
+ int streamType,
+ uint32_t sampleRate,
+ int format,
+ int channelCount,
+ const sp<IMemory>& sharedBuffer,
+ uint32_t flags,
+ callback_t cbf,
+ void* user,
+ int notificationFrames)
+ : mStatus(NO_INIT)
+{
+ mStatus = set(streamType, sampleRate, format, channelCount,
+ 0, flags, cbf, user, notificationFrames, sharedBuffer);
}
AudioTrack::~AudioTrack()
{
+ LOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer());
+
if (mStatus == NO_ERROR) {
- if (mPosition) {
- releaseBuffer(&mAudioBuffer);
- }
- // obtainBuffer() will give up with an error
- mAudioTrack->stop();
+ // Make sure that callback function exits in the case where
+ // it is looping on buffer full condition in obtainBuffer().
+ // Otherwise the callback thread will never exit.
+ stop();
if (mAudioTrackThread != 0) {
+ mCblk->cv.signal();
mAudioTrackThread->requestExitAndWait();
mAudioTrackThread.clear();
}
@@ -98,11 +106,17 @@
uint32_t sampleRate,
int format,
int channelCount,
- int bufferCount,
+ int frameCount,
uint32_t flags,
- callback_t cbf, void* user)
+ callback_t cbf,
+ void* user,
+ int notificationFrames,
+ const sp<IMemory>& sharedBuffer,
+ bool threadCanCallJava)
{
+ LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
+
if (mAudioFlinger != 0) {
LOGE("Track already in use");
return INVALID_OPERATION;
@@ -113,13 +127,26 @@
LOGE("Could not get audioflinger");
return NO_INIT;
}
+ int afSampleRate;
+ if (AudioSystem::getOutputSamplingRate(&afSampleRate) != NO_ERROR) {
+ return NO_INIT;
+ }
+ int afFrameCount;
+ if (AudioSystem::getOutputFrameCount(&afFrameCount) != NO_ERROR) {
+ return NO_INIT;
+ }
+ uint32_t afLatency;
+ if (AudioSystem::getOutputLatency(&afLatency) != NO_ERROR) {
+ return NO_INIT;
+ }
+
// handle default values first.
if (streamType == DEFAULT) {
streamType = MUSIC;
}
if (sampleRate == 0) {
- sampleRate = audioFlinger->sampleRate();
+ sampleRate = afSampleRate;
}
// these below should probably come from the audioFlinger too...
if (format == 0) {
@@ -128,12 +155,10 @@
if (channelCount == 0) {
channelCount = 2;
}
- if (bufferCount == 0) {
- bufferCount = 2;
- }
// validate parameters
- if (format != AudioSystem::PCM_16_BIT) {
+ if (((format != AudioSystem::PCM_8_BIT) || mSharedBuffer != 0) &&
+ (format != AudioSystem::PCM_16_BIT)) {
LOGE("Invalid format");
return BAD_VALUE;
}
@@ -141,17 +166,51 @@
LOGE("Invalid channel number");
return BAD_VALUE;
}
- if (bufferCount < 2) {
- LOGE("Invalid buffer count");
- return BAD_VALUE;
+
+ // Ensure that buffer depth covers at least audio hardware latency
+ uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
+ // When playing from shared buffer, playback will start even if last audioflinger
+ // block is partly filled.
+ if (sharedBuffer != 0 && minBufCount > 1) {
+ minBufCount--;
+ }
+
+ int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
+
+ if (sharedBuffer == 0) {
+ if (frameCount == 0) {
+ frameCount = minFrameCount;
+ }
+ if (notificationFrames == 0) {
+ notificationFrames = frameCount/2;
+ }
+ // Make sure that application is notified with sufficient margin
+ // before underrun
+ if (notificationFrames > frameCount/2) {
+ notificationFrames = frameCount/2;
+ }
+ } else {
+ // Ensure that buffer alignment matches channelcount
+ if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) {
+ LOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount);
+ return BAD_VALUE;
+ }
+ frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t);
+ }
+
+ if (frameCount < minFrameCount) {
+ LOGE("Invalid buffer size: minFrameCount %d, frameCount %d", minFrameCount, frameCount);
+ return BAD_VALUE;
}
// create the track
+ status_t status;
sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
- streamType, sampleRate, format, channelCount, bufferCount, flags);
+ streamType, sampleRate, format, channelCount, frameCount, flags, sharedBuffer, &status);
+
if (track == 0) {
- LOGE("AudioFlinger could not create track");
- return NO_INIT;
+ LOGE("AudioFlinger could not create track, status: %d", status);
+ return status;
}
sp<IMemory> cblk = track->getCblk();
if (cblk == 0) {
@@ -159,7 +218,7 @@
return NO_INIT;
}
if (cbf != 0) {
- mAudioTrackThread = new AudioTrackThread(*this);
+ mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
if (mAudioTrackThread == 0) {
LOGE("Could not create callback thread");
return NO_INIT;
@@ -172,23 +231,34 @@
mAudioTrack = track;
mCblkMemory = cblk;
mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
- mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
+ if (sharedBuffer == 0) {
+ mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
+ } else {
+ mCblk->buffers = sharedBuffer->pointer();
+ }
+ mCblk->out = 1;
mCblk->volume[0] = mCblk->volume[1] = 0x1000;
mVolume[LEFT] = 1.0f;
mVolume[RIGHT] = 1.0f;
mSampleRate = sampleRate;
- mFrameCount = audioFlinger->frameCount();
mStreamType = streamType;
mFormat = format;
- mBufferCount = bufferCount;
+ // Update buffer size in case it has been limited by AudioFlinger during track creation
+ mFrameCount = mCblk->frameCount;
mChannelCount = channelCount;
+ mSharedBuffer = sharedBuffer;
mMuted = false;
mActive = 0;
- mReserved = 0;
mCbf = cbf;
+ mNotificationFrames = notificationFrames;
+ mRemainingFrames = notificationFrames;
mUserData = user;
- mLatency = seconds(mFrameCount) / mSampleRate;
- mPosition = 0;
+ mLatency = afLatency + (1000*mFrameCount) / mSampleRate;
+ mLoopCount = 0;
+ mMarkerPosition = 0;
+ mNewPosition = 0;
+ mUpdatePeriod = 0;
+
return NO_ERROR;
}
@@ -199,7 +269,7 @@
// -------------------------------------------------------------------------
-nsecs_t AudioTrack::latency() const
+uint32_t AudioTrack::latency() const
{
return mLatency;
}
@@ -224,9 +294,19 @@
return mChannelCount;
}
-int AudioTrack::bufferCount() const
+uint32_t AudioTrack::frameCount() const
{
- return mBufferCount;
+ return mFrameCount;
+}
+
+int AudioTrack::frameSize() const
+{
+ return channelCount()*((format() == AudioSystem::PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t));
+}
+
+sp<IMemory>& AudioTrack::sharedBuffer()
+{
+ return mSharedBuffer;
}
// -------------------------------------------------------------------------
@@ -247,6 +327,12 @@
}
if (android_atomic_or(1, &mActive) == 0) {
+ if (mSharedBuffer != 0) {
+ // Force buffer full condition as data is already present in shared memory
+ mCblk->user = mFrameCount;
+ mCblk->flowControlFlag = 0;
+ }
+ mNewPosition = mCblk->server + mUpdatePeriod;
if (t != 0) {
t->run("AudioTrackThread", THREAD_PRIORITY_AUDIO_CLIENT);
} else {
@@ -270,15 +356,20 @@
}
if (android_atomic_and(~1, &mActive) == 1) {
- if (mPosition) {
- releaseBuffer(&mAudioBuffer);
- }
mAudioTrack->stop();
- if (t != 0) {
- t->requestExit();
- } else {
- setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
- }
+ // Cancel loops (If we are in the middle of a loop, playback
+ // would not stop until loopCount reaches 0).
+ setLoop(0, 0, 0);
+ // Force flush if a shared buffer is used otherwise audioflinger
+ // will not stop before end of buffer is reached.
+ if (mSharedBuffer != 0) {
+ flush();
+ }
+ if (t != 0) {
+ t->requestExit();
+ } else {
+ setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
+ }
}
if (t != 0) {
@@ -294,6 +385,7 @@
void AudioTrack::flush()
{
LOGV("flush");
+
if (!mActive) {
mCblk->lock.lock();
mAudioTrack->flush();
@@ -341,7 +433,16 @@
void AudioTrack::setSampleRate(int rate)
{
+ int afSamplingRate;
+
+ if (AudioSystem::getOutputSamplingRate(&afSamplingRate) != NO_ERROR) {
+ return;
+ }
+ // Resampler implementation limits input sampling rate to 2 x output sampling rate.
+ if (rate > afSamplingRate*2) rate = afSamplingRate*2;
+
if (rate > MAX_SAMPLE_RATE) rate = MAX_SAMPLE_RATE;
+
mCblk->sampleRate = rate;
}
@@ -350,6 +451,129 @@
return uint32_t(mCblk->sampleRate);
}
+status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
+{
+ audio_track_cblk_t* cblk = mCblk;
+
+
+ Mutex::Autolock _l(cblk->lock);
+
+ if (loopCount == 0) {
+ cblk->loopStart = UINT_MAX;
+ cblk->loopEnd = UINT_MAX;
+ cblk->loopCount = 0;
+ mLoopCount = 0;
+ return NO_ERROR;
+ }
+
+ if (loopStart >= loopEnd ||
+ loopStart < cblk->user ||
+ loopEnd - loopStart > mFrameCount) {
+ LOGW("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, mFrameCount, cblk->user);
+ return BAD_VALUE;
+ }
+ // TODO handle shared buffer here: limit loop end to framecount
+
+ cblk->loopStart = loopStart;
+ cblk->loopEnd = loopEnd;
+ cblk->loopCount = loopCount;
+ mLoopCount = loopCount;
+
+ return NO_ERROR;
+}
+
+status_t AudioTrack::getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount)
+{
+ if (loopStart != 0) {
+ *loopStart = mCblk->loopStart;
+ }
+ if (loopEnd != 0) {
+ *loopEnd = mCblk->loopEnd;
+ }
+ if (loopCount != 0) {
+ if (mCblk->loopCount < 0) {
+ *loopCount = -1;
+ } else {
+ *loopCount = mCblk->loopCount;
+ }
+ }
+
+ return NO_ERROR;
+}
+
+status_t AudioTrack::setMarkerPosition(uint32_t marker)
+{
+ if (mCbf == 0) return INVALID_OPERATION;
+
+ mMarkerPosition = marker;
+
+ return NO_ERROR;
+}
+
+status_t AudioTrack::getMarkerPosition(uint32_t *marker)
+{
+ if (marker == 0) return BAD_VALUE;
+
+ *marker = mMarkerPosition;
+
+ return NO_ERROR;
+}
+
+status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
+{
+ if (mCbf == 0) return INVALID_OPERATION;
+
+ uint32_t curPosition;
+ getPosition(&curPosition);
+ mNewPosition = curPosition + updatePeriod;
+ mUpdatePeriod = updatePeriod;
+
+ return NO_ERROR;
+}
+
+status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod)
+{
+ if (updatePeriod == 0) return BAD_VALUE;
+
+ *updatePeriod = mUpdatePeriod;
+
+ return NO_ERROR;
+}
+
+status_t AudioTrack::setPosition(uint32_t position)
+{
+ Mutex::Autolock _l(mCblk->lock);
+
+ if (!stopped()) return INVALID_OPERATION;
+
+ if (position > mCblk->user) return BAD_VALUE;
+
+ mCblk->server = position;
+ mCblk->forceReady = 1;
+
+ return NO_ERROR;
+}
+
+status_t AudioTrack::getPosition(uint32_t *position)
+{
+ if (position == 0) return BAD_VALUE;
+
+ *position = mCblk->server;
+
+ return NO_ERROR;
+}
+
+status_t AudioTrack::reload()
+{
+ if (!stopped()) return INVALID_OPERATION;
+
+ flush();
+
+ mCblk->stepUser(mFrameCount);
+
+ return NO_ERROR;
+}
+
// -------------------------------------------------------------------------
status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, bool blocking)
@@ -358,21 +582,17 @@
int timeout = 0;
status_t result;
audio_track_cblk_t* cblk = mCblk;
+ uint32_t framesReq = audioBuffer->frameCount;
- uint32_t u = cblk->user;
- uint32_t u_seq = u & audio_track_cblk_t::SEQUENCE_MASK;
- uint32_t u_buf = u & audio_track_cblk_t::BUFFER_MASK;
+ audioBuffer->frameCount = 0;
+ audioBuffer->size = 0;
- uint32_t s = cblk->server;
- uint32_t s_seq = s & audio_track_cblk_t::SEQUENCE_MASK;
- uint32_t s_buf = s & audio_track_cblk_t::BUFFER_MASK;
+ uint32_t framesAvail = cblk->framesAvailable();
- LOGW_IF(u_seq < s_seq, "user doesn't fill buffers fast enough");
-
- if (u_seq > s_seq && u_buf == s_buf) {
+ if (framesAvail == 0) {
Mutex::Autolock _l(cblk->lock);
goto start_loop_here;
- while (u_seq > s_seq && u_buf == s_buf) {
+ while (framesAvail == 0) {
active = mActive;
if (UNLIKELY(!active)) {
LOGV("Not active and NO_MORE_BUFFERS");
@@ -384,89 +604,101 @@
result = cblk->cv.waitRelative(cblk->lock, seconds(1));
if (__builtin_expect(result!=NO_ERROR, false)) {
LOGW( "obtainBuffer timed out (is the CPU pegged?) "
- "user=%08x, server=%08x", u, s);
+ "user=%08x, server=%08x", cblk->user, cblk->server);
mAudioTrack->start(); // FIXME: Wake up audioflinger
timeout = 1;
}
- // Read user count in case a flush has reset while we where waiting on cv.
- u = cblk->user;
- u_seq = u & audio_track_cblk_t::SEQUENCE_MASK;
- u_buf = u & audio_track_cblk_t::BUFFER_MASK;
-
// read the server count again
start_loop_here:
- s = cblk->server;
- s_seq = s & audio_track_cblk_t::SEQUENCE_MASK;
- s_buf = s & audio_track_cblk_t::BUFFER_MASK;
+ framesAvail = cblk->framesAvailable_l();
}
}
+ if (framesReq > framesAvail) {
+ framesReq = framesAvail;
+ }
+
+ uint32_t u = cblk->user;
+ uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
+
+ if (u + framesReq > bufferEnd) {
+ framesReq = bufferEnd - u;
+ }
+
LOGW_IF(timeout,
"*** SERIOUS WARNING *** obtainBuffer() timed out "
"but didn't need to be locked. We recovered, but "
- "this shouldn't happen (user=%08x, server=%08x)", u, s);
+ "this shouldn't happen (user=%08x, server=%08x)", cblk->user, cblk->server);
audioBuffer->flags = mMuted ? Buffer::MUTE : 0;
audioBuffer->channelCount= mChannelCount;
- audioBuffer->format = mFormat;
- audioBuffer->frameCount = mFrameCount;
- audioBuffer->size = cblk->size;
- audioBuffer->raw = (int8_t *)cblk->buffer(u_buf);
+ audioBuffer->format = AudioSystem::PCM_16_BIT;
+ audioBuffer->frameCount = framesReq;
+ audioBuffer->size = framesReq*mChannelCount*sizeof(int16_t);
+ audioBuffer->raw = (int8_t *)cblk->buffer(u);
active = mActive;
return active ? status_t(NO_ERROR) : status_t(STOPPED);
}
void AudioTrack::releaseBuffer(Buffer* audioBuffer)
{
- // next buffer...
- if (UNLIKELY(mPosition)) {
- // clean the remaining part of the buffer
- size_t capacity = mAudioBuffer.size - mPosition;
- memset(mAudioBuffer.i8 + mPosition, 0, capacity);
- mPosition = 0;
- }
audio_track_cblk_t* cblk = mCblk;
- cblk->stepUser(mBufferCount);
+ cblk->stepUser(audioBuffer->frameCount);
}
// -------------------------------------------------------------------------
ssize_t AudioTrack::write(const void* buffer, size_t userSize)
{
+
+ if (mSharedBuffer != 0) return INVALID_OPERATION;
+
if (ssize_t(userSize) < 0) {
// sanity-check. user is most-likely passing an error code.
- LOGE("AudioTrack::write(buffer=%p, size=%u (%d)",
+ LOGE("AudioTrack::write(buffer=%p, size=%u (%d)",
buffer, userSize, userSize);
return BAD_VALUE;
}
LOGV("write %d bytes, mActive=%d", userSize, mActive);
+
ssize_t written = 0;
+ const int8_t *src = (const int8_t *)buffer;
+ Buffer audioBuffer;
+
do {
- if (mPosition == 0) {
- status_t err = obtainBuffer(&mAudioBuffer, true);
- if (err < 0) {
- // out of buffers, return #bytes written
- if (err == status_t(NO_MORE_BUFFERS))
- break;
- return ssize_t(err);
- }
+ audioBuffer.frameCount = userSize/mChannelCount;
+ if (mFormat == AudioSystem::PCM_16_BIT) {
+ audioBuffer.frameCount >>= 1;
}
- size_t capacity = mAudioBuffer.size - mPosition;
- size_t toWrite = userSize < capacity ? userSize : capacity;
+ status_t err = obtainBuffer(&audioBuffer, true);
+ if (err < 0) {
+ // out of buffers, return #bytes written
+ if (err == status_t(NO_MORE_BUFFERS))
+ break;
+ return ssize_t(err);
+ }
- memcpy(mAudioBuffer.i8 + mPosition, buffer, toWrite);
- buffer = static_cast<const int8_t*>(buffer) + toWrite;
- mPosition += toWrite;
+ size_t toWrite;
+ if (mFormat == AudioSystem::PCM_8_BIT) {
+ // Divide capacity by 2 to take expansion into account
+ toWrite = audioBuffer.size>>1;
+ // 8 to 16 bit conversion
+ int count = toWrite;
+ int16_t *dst = (int16_t *)(audioBuffer.i8);
+ while(count--) {
+ *dst++ = (int16_t)(*src++^0x80) << 8;
+ }
+ }else {
+ toWrite = audioBuffer.size;
+ memcpy(audioBuffer.i8, src, toWrite);
+ src += toWrite;
+ }
userSize -= toWrite;
- capacity -= toWrite;
written += toWrite;
- if (capacity == 0) {
- mPosition = 0;
- releaseBuffer(&mAudioBuffer);
- }
+ releaseBuffer(&audioBuffer);
} while (userSize);
return written;
@@ -477,16 +709,115 @@
bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
{
Buffer audioBuffer;
+ uint32_t frames;
+ size_t writtenSize = 0;
- status_t err = obtainBuffer(&audioBuffer, true);
- if (err < NO_ERROR) {
- LOGE("Error obtaining an audio buffer, giving up.");
- return false;
+ // Manage underrun callback
+ if (mActive && (mCblk->framesReady() == 0)) {
+ LOGV("Underrun user: %x, server: %x, flowControlFlag %d", mCblk->user, mCblk->server, mCblk->flowControlFlag);
+ if (mCblk->flowControlFlag == 0) {
+ mCbf(EVENT_UNDERRUN, mUserData, 0);
+ if (mCblk->server == mCblk->frameCount) {
+ mCbf(EVENT_BUFFER_END, mUserData, 0);
+ }
+ mCblk->flowControlFlag = 1;
+ if (mSharedBuffer != 0) return false;
+ }
}
- if (err == status_t(STOPPED)) return false;
- mCbf(mUserData, audioBuffer);
- releaseBuffer(&audioBuffer);
+
+ // Manage loop end callback
+ while (mLoopCount > mCblk->loopCount) {
+ int loopCount = -1;
+ mLoopCount--;
+ if (mLoopCount >= 0) loopCount = mLoopCount;
+ mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount);
+ }
+
+ // Manage marker callback
+ if(mMarkerPosition > 0) {
+ if (mCblk->server >= mMarkerPosition) {
+ mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition);
+ mMarkerPosition = 0;
+ }
+ }
+
+ // Manage new position callback
+ if(mUpdatePeriod > 0) {
+ while (mCblk->server >= mNewPosition) {
+ mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition);
+ mNewPosition += mUpdatePeriod;
+ }
+ }
+
+ // If Shared buffer is used, no data is requested from client.
+ if (mSharedBuffer != 0) {
+ frames = 0;
+ } else {
+ frames = mRemainingFrames;
+ }
+
+ do {
+
+ audioBuffer.frameCount = frames;
+
+ status_t err = obtainBuffer(&audioBuffer, false);
+ if (err < NO_ERROR) {
+ if (err != WOULD_BLOCK) {
+ LOGE("Error obtaining an audio buffer, giving up.");
+ return false;
+ }
+ }
+ if (err == status_t(STOPPED)) return false;
+
+ if (audioBuffer.size == 0) break;
+
+ // Divide buffer size by 2 to take into account the expansion
+ // due to 8 to 16 bit conversion: the callback must fill only half
+ // of the destination buffer
+ if (mFormat == AudioSystem::PCM_8_BIT) {
+ audioBuffer.size >>= 1;
+ }
+
+ size_t reqSize = audioBuffer.size;
+ mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
+ writtenSize = audioBuffer.size;
+
+ // Sanity check on returned size
+ if (ssize_t(writtenSize) <= 0) break;
+ if (writtenSize > reqSize) writtenSize = reqSize;
+
+ if (mFormat == AudioSystem::PCM_8_BIT) {
+ // 8 to 16 bit conversion
+ const int8_t *src = audioBuffer.i8 + writtenSize-1;
+ int count = writtenSize;
+ int16_t *dst = audioBuffer.i16 + writtenSize-1;
+ while(count--) {
+ *dst-- = (int16_t)(*src--^0x80) << 8;
+ }
+ writtenSize <<= 1;
+ }
+
+ audioBuffer.size = writtenSize;
+ audioBuffer.frameCount = writtenSize/mChannelCount/sizeof(int16_t);
+ frames -= audioBuffer.frameCount;
+
+ releaseBuffer(&audioBuffer);
+ }
+ while (frames);
+
+ // If no data was written, it is likely that obtainBuffer() did
+ // not find room in PCM buffer: we release the processor for
+ // a few millisecond before polling again for available room.
+ if (writtenSize == 0) {
+ usleep(5000);
+ }
+
+ if (frames == 0) {
+ mRemainingFrames = mNotificationFrames;
+ } else {
+ mRemainingFrames = frames;
+ }
return true;
}
@@ -500,11 +831,11 @@
result.append(" AudioTrack::dump\n");
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), buffer count(%d)\n", mFormat, mChannelCount, mFrameCount, mBufferCount);
+ snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mFrameCount);
result.append(buffer);
- snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d), reserved(%d)\n", mSampleRate, mStatus, mMuted, mReserved);
+ snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n", mSampleRate, mStatus, mMuted);
result.append(buffer);
- snprintf(buffer, 255, " active(%d), latency (%lld), position(%d)\n", mActive, mLatency, mPosition);
+ snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency);
result.append(buffer);
::write(fd, result.string(), result.size());
return NO_ERROR;
@@ -512,8 +843,8 @@
// =========================================================================
-AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver)
- : Thread(false), mReceiver(receiver)
+AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava)
+ : Thread(bCanCallJava), mReceiver(receiver)
{
}
@@ -534,25 +865,35 @@
// =========================================================================
audio_track_cblk_t::audio_track_cblk_t()
- : user(0), server(0), volumeLR(0), buffers(0), size(0)
+ : user(0), server(0), userBase(0), serverBase(0), buffers(0), frameCount(0),
+ loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), volumeLR(0), flowControlFlag(1), forceReady(0)
{
}
-uint32_t audio_track_cblk_t::stepUser(int bufferCount)
+uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount)
{
uint32_t u = this->user;
- uint32_t u_seq = u & audio_track_cblk_t::SEQUENCE_MASK;
- uint32_t u_buf = u & audio_track_cblk_t::BUFFER_MASK;
- if (++u_buf >= uint32_t(bufferCount)) {
- u_seq += 0x100;
- u_buf = 0;
+
+ u += frameCount;
+ // Ensure that user is never ahead of server for AudioRecord
+ if (!out && u > this->server) {
+ LOGW("stepServer occured after track reset");
+ u = this->server;
}
- u = u_seq | u_buf;
- this->user = u;
+
+ if (u >= userBase + this->frameCount) {
+ userBase += this->frameCount;
+ }
+
+ this->user = u;
+
+ // Clear flow control error condition as new data has been written/read to/from buffer.
+ flowControlFlag = 0;
+
return u;
}
-bool audio_track_cblk_t::stepServer(int bufferCount)
+bool audio_track_cblk_t::stepServer(uint32_t frameCount)
{
// the code below simulates lock-with-timeout
// we MUST do this to protect the AudioFlinger server
@@ -570,24 +911,83 @@
}
uint32_t s = this->server;
- uint32_t s_seq = s & audio_track_cblk_t::SEQUENCE_MASK;
- uint32_t s_buf = s & audio_track_cblk_t::BUFFER_MASK;
- s_buf++;
- if (s_buf >= uint32_t(bufferCount)) {
- s_seq += 0x100;
- s_buf = 0;
- }
- s = s_seq | s_buf;
- this->server = s;
+ s += frameCount;
+ // It is possible that we receive a flush()
+ // while the mixer is processing a block: in this case,
+ // stepServer() is called After the flush() has reset u & s and
+ // we have s > u
+ if (out && s > this->user) {
+ LOGW("stepServer occured after track reset");
+ s = this->user;
+ }
+
+ if (s >= loopEnd) {
+ LOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd);
+ s = loopStart;
+ if (--loopCount == 0) {
+ loopEnd = UINT_MAX;
+ loopStart = UINT_MAX;
+ }
+ }
+ if (s >= serverBase + this->frameCount) {
+ serverBase += this->frameCount;
+ }
+
+ this->server = s;
+
cv.signal();
lock.unlock();
return true;
}
-void* audio_track_cblk_t::buffer(int id) const
+void* audio_track_cblk_t::buffer(uint32_t offset) const
{
- return (char*)this->buffers + id * this->size;
+ return (int16_t *)this->buffers + (offset-userBase)*this->channels;
+}
+
+uint32_t audio_track_cblk_t::framesAvailable()
+{
+ Mutex::Autolock _l(lock);
+ return framesAvailable_l();
+}
+
+uint32_t audio_track_cblk_t::framesAvailable_l()
+{
+ uint32_t u = this->user;
+ uint32_t s = this->server;
+
+ if (out) {
+ if (u < loopEnd) {
+ return s + frameCount - u;
+ } else {
+ uint32_t limit = (s < loopStart) ? s : loopStart;
+ return limit + frameCount - u;
+ }
+ } else {
+ return frameCount + u - s;
+ }
+}
+
+uint32_t audio_track_cblk_t::framesReady()
+{
+ uint32_t u = this->user;
+ uint32_t s = this->server;
+
+ if (out) {
+ if (u < loopEnd) {
+ return u - s;
+ } else {
+ Mutex::Autolock _l(lock);
+ if (loopCount >= 0) {
+ return (loopEnd - loopStart)*loopCount + u - s;
+ } else {
+ return UINT_MAX;
+ }
+ }
+ } else {
+ return s - u;
+ }
}
// -------------------------------------------------------------------------