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-rw-r--r--media/libmedia/AudioRecord.cpp20
-rw-r--r--media/libmedia/AudioSystem.cpp32
-rw-r--r--media/libmedia/AudioTrack.cpp54
-rw-r--r--media/libmedia/IAudioFlinger.cpp24
-rw-r--r--media/libmedia/SoundPool.cpp4
5 files changed, 69 insertions, 65 deletions
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 062f546..0587651 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -36,7 +36,7 @@ namespace android {
// static
status_t AudioRecord::getMinFrameCount(
- int* frameCount,
+ size_t* frameCount,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask)
@@ -54,7 +54,7 @@ status_t AudioRecord::getMinFrameCount(
}
if (size == 0) {
- ALOGE("Unsupported configuration: sampleRate %d, format %d, channelMask %#x",
+ ALOGE("Unsupported configuration: sampleRate %u, format %d, channelMask %#x",
sampleRate, format, channelMask);
return BAD_VALUE;
}
@@ -119,15 +119,21 @@ status_t AudioRecord::set(
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- int frameCount,
+ int frameCountInt,
callback_t cbf,
void* user,
int notificationFrames,
bool threadCanCallJava,
int sessionId)
{
+ // FIXME "int" here is legacy and will be replaced by size_t later
+ if (frameCountInt < 0) {
+ ALOGE("Invalid frame count %d", frameCountInt);
+ return BAD_VALUE;
+ }
+ size_t frameCount = frameCountInt;
- ALOGV("set(): sampleRate %d, channelMask %#x, frameCount %d",sampleRate, channelMask,
+ ALOGV("set(): sampleRate %u, channelMask %#x, frameCount %u", sampleRate, channelMask,
frameCount);
AutoMutex lock(mLock);
@@ -177,7 +183,7 @@ status_t AudioRecord::set(
}
// validate framecount
- int minFrameCount = 0;
+ size_t minFrameCount = 0;
status_t status = getMinFrameCount(&minFrameCount, sampleRate, format, channelMask);
if (status != NO_ERROR) {
return status;
@@ -260,7 +266,7 @@ int AudioRecord::channelCount() const
return mChannelCount;
}
-uint32_t AudioRecord::frameCount() const
+size_t AudioRecord::frameCount() const
{
return mFrameCount;
}
@@ -427,7 +433,7 @@ status_t AudioRecord::openRecord_l(
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- int frameCount,
+ size_t frameCount,
audio_io_handle_t input)
{
status_t status;
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 767c452..028e4a3 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -205,12 +205,7 @@ int AudioSystem::logToLinear(float volume)
return volume ? 100 - int(dBConvertInverse * log(volume) + 0.5) : 0;
}
-// DEPRECATED
-status_t AudioSystem::getOutputSamplingRate(int* samplingRate, int streamType) {
- return getOutputSamplingRate(samplingRate, (audio_stream_type_t)streamType);
-}
-
-status_t AudioSystem::getOutputSamplingRate(int* samplingRate, audio_stream_type_t streamType)
+status_t AudioSystem::getOutputSamplingRate(uint32_t* samplingRate, audio_stream_type_t streamType)
{
audio_io_handle_t output;
@@ -228,7 +223,7 @@ status_t AudioSystem::getOutputSamplingRate(int* samplingRate, audio_stream_type
status_t AudioSystem::getSamplingRate(audio_io_handle_t output,
audio_stream_type_t streamType,
- int* samplingRate)
+ uint32_t* samplingRate)
{
OutputDescriptor *outputDesc;
@@ -246,18 +241,13 @@ status_t AudioSystem::getSamplingRate(audio_io_handle_t output,
gLock.unlock();
}
- ALOGV("getSamplingRate() streamType %d, output %d, sampling rate %d", streamType, output,
+ ALOGV("getSamplingRate() streamType %d, output %d, sampling rate %u", streamType, output,
*samplingRate);
return NO_ERROR;
}
-// DEPRECATED
-status_t AudioSystem::getOutputFrameCount(int* frameCount, int streamType) {
- return getOutputFrameCount(frameCount, (audio_stream_type_t)streamType);
-}
-
-status_t AudioSystem::getOutputFrameCount(int* frameCount, audio_stream_type_t streamType)
+status_t AudioSystem::getOutputFrameCount(size_t* frameCount, audio_stream_type_t streamType)
{
audio_io_handle_t output;
@@ -275,7 +265,7 @@ status_t AudioSystem::getOutputFrameCount(int* frameCount, audio_stream_type_t s
status_t AudioSystem::getFrameCount(audio_io_handle_t output,
audio_stream_type_t streamType,
- int* frameCount)
+ size_t* frameCount)
{
OutputDescriptor *outputDesc;
@@ -371,7 +361,7 @@ status_t AudioSystem::setVoiceVolume(float value)
return af->setVoiceVolume(value);
}
-status_t AudioSystem::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
+status_t AudioSystem::getRenderPosition(size_t *halFrames, size_t *dspFrames,
audio_stream_type_t stream)
{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
@@ -384,7 +374,7 @@ status_t AudioSystem::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames
return af->getRenderPosition(halFrames, dspFrames, getOutput(stream));
}
-unsigned int AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) {
+size_t AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) {
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
unsigned int result = 0;
if (af == 0) return result;
@@ -452,7 +442,7 @@ void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, audio_io_handle
OutputDescriptor *outputDesc = new OutputDescriptor(*desc);
gOutputs.add(ioHandle, outputDesc);
- ALOGV("ioConfigChanged() new output samplingRate %d, format %d channels %#x frameCount %d "
+ ALOGV("ioConfigChanged() new output samplingRate %u, format %d channels %#x frameCount %u "
"latency %d",
outputDesc->samplingRate, outputDesc->format, outputDesc->channels,
outputDesc->frameCount, outputDesc->latency);
@@ -476,7 +466,7 @@ void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, audio_io_handle
if (param2 == NULL) break;
desc = (const OutputDescriptor *)param2;
- ALOGV("ioConfigChanged() new config for output %d samplingRate %d, format %d channels %#x "
+ ALOGV("ioConfigChanged() new config for output %d samplingRate %u, format %d channels %#x "
"frameCount %d latency %d",
ioHandle, desc->samplingRate, desc->format,
desc->channels, desc->frameCount, desc->latency);
@@ -750,14 +740,14 @@ status_t AudioSystem::isSourceActive(audio_source_t stream, bool* state)
return NO_ERROR;
}
-int32_t AudioSystem::getPrimaryOutputSamplingRate()
+uint32_t AudioSystem::getPrimaryOutputSamplingRate()
{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return 0;
return af->getPrimaryOutputSamplingRate();
}
-int32_t AudioSystem::getPrimaryOutputFrameCount()
+size_t AudioSystem::getPrimaryOutputFrameCount()
{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return 0;
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 4a4759e..979ee37 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -50,7 +50,7 @@ namespace android {
// static
status_t AudioTrack::getMinFrameCount(
- int* frameCount,
+ size_t* frameCount,
audio_stream_type_t streamType,
uint32_t sampleRate)
{
@@ -65,11 +65,11 @@ status_t AudioTrack::getMinFrameCount(
// audio_format_t format
// audio_channel_mask_t channelMask
// audio_output_flags_t flags
- int afSampleRate;
+ uint32_t afSampleRate;
if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
return NO_INIT;
}
- int afFrameCount;
+ size_t afFrameCount;
if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
return NO_INIT;
}
@@ -166,7 +166,7 @@ status_t AudioTrack::set(
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- int frameCount,
+ int frameCountInt,
audio_output_flags_t flags,
callback_t cbf,
void* user,
@@ -175,11 +175,17 @@ status_t AudioTrack::set(
bool threadCanCallJava,
int sessionId)
{
+ // FIXME "int" here is legacy and will be replaced by size_t later
+ if (frameCountInt < 0) {
+ ALOGE("Invalid frame count %d", frameCountInt);
+ return BAD_VALUE;
+ }
+ size_t frameCount = frameCountInt;
ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(),
sharedBuffer->size());
- ALOGV("set() streamType %d frameCount %d flags %04x", streamType, frameCount, flags);
+ ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags);
AutoMutex lock(mLock);
if (mAudioTrack != 0) {
@@ -193,7 +199,7 @@ status_t AudioTrack::set(
}
if (sampleRate == 0) {
- int afSampleRate;
+ uint32_t afSampleRate;
if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
return NO_INIT;
}
@@ -336,7 +342,7 @@ int AudioTrack::channelCount() const
return mChannelCount;
}
-uint32_t AudioTrack::frameCount() const
+size_t AudioTrack::frameCount() const
{
return mCblk->frameCount;
}
@@ -390,7 +396,7 @@ void AudioTrack::start()
}
if (cblk->flags & CBLK_INVALID) {
audio_track_cblk_t* temp = cblk;
- status = restoreTrack_l(temp, true);
+ status = restoreTrack_l(temp, true /*fromStart*/);
cblk = temp;
}
cblk->lock.unlock();
@@ -535,9 +541,9 @@ void AudioTrack::getAuxEffectSendLevel(float* level) const
}
}
-status_t AudioTrack::setSampleRate(int rate)
+status_t AudioTrack::setSampleRate(uint32_t rate)
{
- int afSamplingRate;
+ uint32_t afSamplingRate;
if (mIsTimed) {
return INVALID_OPERATION;
@@ -547,7 +553,7 @@ status_t AudioTrack::setSampleRate(int rate)
return NO_INIT;
}
// Resampler implementation limits input sampling rate to 2 x output sampling rate.
- if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE;
+ if (rate == 0 || rate > afSamplingRate*2 ) return BAD_VALUE;
AutoMutex lock(mLock);
mCblk->sampleRate = rate;
@@ -557,7 +563,7 @@ status_t AudioTrack::setSampleRate(int rate)
uint32_t AudioTrack::getSampleRate() const
{
if (mIsTimed) {
- return INVALID_OPERATION;
+ return 0;
}
AutoMutex lock(mLock);
@@ -730,7 +736,7 @@ status_t AudioTrack::createTrack_l(
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- int frameCount,
+ size_t frameCount,
audio_output_flags_t flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output)
@@ -770,7 +776,7 @@ status_t AudioTrack::createTrack_l(
// Same comment as below about ignoring frameCount parameter for set()
frameCount = sharedBuffer->size();
} else if (frameCount == 0) {
- int afFrameCount;
+ size_t afFrameCount;
if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) {
return NO_INIT;
}
@@ -802,11 +808,11 @@ status_t AudioTrack::createTrack_l(
} else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
// FIXME move these calculations and associated checks to server
- int afSampleRate;
+ uint32_t afSampleRate;
if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) {
return NO_INIT;
}
- int afFrameCount;
+ size_t afFrameCount;
if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) {
return NO_INIT;
}
@@ -815,8 +821,8 @@ status_t AudioTrack::createTrack_l(
uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
if (minBufCount < 2) minBufCount = 2;
- int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
- ALOGV("minFrameCount: %d, afFrameCount=%d, minBufCount=%d, sampleRate=%d, afSampleRate=%d"
+ size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
+ ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u"
", afLatency=%d",
minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency);
@@ -828,7 +834,7 @@ status_t AudioTrack::createTrack_l(
}
// Make sure that application is notified with sufficient margin
// before underrun
- if (mNotificationFramesAct > (uint32_t)frameCount/2) {
+ if (mNotificationFramesAct > frameCount/2) {
mNotificationFramesAct = frameCount/2;
}
if (frameCount < minFrameCount) {
@@ -858,7 +864,9 @@ status_t AudioTrack::createTrack_l(
sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
streamType,
sampleRate,
- format,
+ // AudioFlinger only sees 16-bit PCM
+ format == AUDIO_FORMAT_PCM_8_BIT ?
+ AUDIO_FORMAT_PCM_16_BIT : format,
channelMask,
frameCount,
&trackFlags,
@@ -988,7 +996,7 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
android_atomic_or(CBLK_INVALID, &cblk->flags);
create_new_track:
audio_track_cblk_t* temp = cblk;
- result = restoreTrack_l(temp, false);
+ result = restoreTrack_l(temp, false /*fromStart*/);
cblk = temp;
}
if (result != NO_ERROR) {
@@ -1147,7 +1155,7 @@ status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer)
if (cblk->flags & CBLK_INVALID) {
cblk->lock.lock();
audio_track_cblk_t* temp = cblk;
- result = restoreTrack_l(temp, false);
+ result = restoreTrack_l(temp, false /*fromStart*/);
cblk = temp;
cblk->lock.unlock();
@@ -1423,7 +1431,7 @@ status_t AudioTrack::dump(int fd, const Vector<String16>& args) const
snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat,
mChannelCount, cblk->frameCount);
result.append(buffer);
- snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n",
+ snprintf(buffer, 255, " sample rate(%u), status(%d), muted(%d)\n",
(cblk == 0) ? 0 : cblk->sampleRate, mStatus, mMuted);
result.append(buffer);
snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency);
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index bb936ec..79c3361 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -89,7 +89,7 @@ public:
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- int frameCount,
+ size_t frameCount,
track_flags_t *flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output,
@@ -106,7 +106,7 @@ public:
data.writeInt32(format);
data.writeInt32(channelMask);
data.writeInt32(frameCount);
- track_flags_t lFlags = flags != NULL ? *flags : TRACK_DEFAULT;
+ track_flags_t lFlags = flags != NULL ? *flags : (track_flags_t) TRACK_DEFAULT;
data.writeInt32(lFlags);
data.writeStrongBinder(sharedBuffer->asBinder());
data.writeInt32((int32_t) output);
@@ -143,7 +143,7 @@ public:
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- int frameCount,
+ size_t frameCount,
track_flags_t flags,
pid_t tid,
int *sessionId,
@@ -527,7 +527,7 @@ public:
return status;
}
- virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const
+ virtual size_t getInputFramesLost(audio_io_handle_t ioHandle) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -695,7 +695,7 @@ public:
return (audio_module_handle_t) reply.readInt32();
}
- virtual int32_t getPrimaryOutputSamplingRate()
+ virtual uint32_t getPrimaryOutputSamplingRate()
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -703,7 +703,7 @@ public:
return reply.readInt32();
}
- virtual int32_t getPrimaryOutputFrameCount()
+ virtual size_t getPrimaryOutputFrameCount()
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -728,7 +728,7 @@ status_t BnAudioFlinger::onTransact(
uint32_t sampleRate = data.readInt32();
audio_format_t format = (audio_format_t) data.readInt32();
audio_channel_mask_t channelMask = data.readInt32();
- size_t bufferCount = data.readInt32();
+ size_t frameCount = data.readInt32();
track_flags_t flags = (track_flags_t) data.readInt32();
sp<IMemory> buffer = interface_cast<IMemory>(data.readStrongBinder());
audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
@@ -737,7 +737,7 @@ status_t BnAudioFlinger::onTransact(
status_t status;
sp<IAudioTrack> track = createTrack(pid,
(audio_stream_type_t) streamType, sampleRate, format,
- channelMask, bufferCount, &flags, buffer, output, tid, &sessionId, &status);
+ channelMask, frameCount, &flags, buffer, output, tid, &sessionId, &status);
reply->writeInt32(flags);
reply->writeInt32(sessionId);
reply->writeInt32(status);
@@ -751,13 +751,13 @@ status_t BnAudioFlinger::onTransact(
uint32_t sampleRate = data.readInt32();
audio_format_t format = (audio_format_t) data.readInt32();
audio_channel_mask_t channelMask = data.readInt32();
- size_t bufferCount = data.readInt32();
+ size_t frameCount = data.readInt32();
track_flags_t flags = (track_flags_t) data.readInt32();
pid_t tid = (pid_t) data.readInt32();
int sessionId = data.readInt32();
status_t status;
sp<IAudioRecord> record = openRecord(pid, input,
- sampleRate, format, channelMask, bufferCount, flags, tid, &sessionId, &status);
+ sampleRate, format, channelMask, frameCount, flags, tid, &sessionId, &status);
reply->writeInt32(sessionId);
reply->writeInt32(status);
reply->writeStrongBinder(record->asBinder());
@@ -972,8 +972,8 @@ status_t BnAudioFlinger::onTransact(
case GET_RENDER_POSITION: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
- uint32_t halFrames;
- uint32_t dspFrames;
+ size_t halFrames;
+ size_t dspFrames;
status_t status = getRenderPosition(&halFrames, &dspFrames, output);
reply->writeInt32(status);
if (status == NO_ERROR) {
diff --git a/media/libmedia/SoundPool.cpp b/media/libmedia/SoundPool.cpp
index abc8899..204e0ce 100644
--- a/media/libmedia/SoundPool.cpp
+++ b/media/libmedia/SoundPool.cpp
@@ -568,8 +568,8 @@ void SoundChannel::play(const sp<Sample>& sample, int nextChannelID, float leftV
}
// initialize track
- int afFrameCount;
- int afSampleRate;
+ size_t afFrameCount;
+ uint32_t afSampleRate;
audio_stream_type_t streamType = mSoundPool->streamType();
if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
afFrameCount = kDefaultFrameCount;
generated by cgit v1.1 at 2024-07-26 07:56:26 +0000