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-rw-r--r--media/libmedia/IAudioFlinger.cpp50
-rw-r--r--services/audioflinger/AudioFlinger.cpp2
-rw-r--r--services/audioflinger/RecordTracks.h1
-rw-r--r--services/audioflinger/Threads.cpp11
-rw-r--r--services/audioflinger/Threads.h2
-rw-r--r--services/audioflinger/Tracks.cpp4
6 files changed, 45 insertions, 25 deletions
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index be818c6..22ad453 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -132,7 +132,7 @@ public:
lStatus = reply.readInt32();
track = interface_cast<IAudioTrack>(reply.readStrongBinder());
}
- if (status) {
+ if (status != NULL) {
*status = lStatus;
}
return track;
@@ -180,7 +180,7 @@ public:
lStatus = reply.readInt32();
record = interface_cast<IAudioRecord>(reply.readStrongBinder());
}
- if (status) {
+ if (status != NULL) {
*status = lStatus;
}
return record;
@@ -397,15 +397,25 @@ public:
audio_io_handle_t output = (audio_io_handle_t) reply.readInt32();
ALOGV("openOutput() returned output, %d", output);
devices = (audio_devices_t)reply.readInt32();
- if (pDevices != NULL) *pDevices = devices;
+ if (pDevices != NULL) {
+ *pDevices = devices;
+ }
samplingRate = reply.readInt32();
- if (pSamplingRate != NULL) *pSamplingRate = samplingRate;
+ if (pSamplingRate != NULL) {
+ *pSamplingRate = samplingRate;
+ }
format = (audio_format_t) reply.readInt32();
- if (pFormat != NULL) *pFormat = format;
+ if (pFormat != NULL) {
+ *pFormat = format;
+ }
channelMask = (audio_channel_mask_t)reply.readInt32();
- if (pChannelMask != NULL) *pChannelMask = channelMask;
+ if (pChannelMask != NULL) {
+ *pChannelMask = channelMask;
+ }
latency = reply.readInt32();
- if (pLatencyMs != NULL) *pLatencyMs = latency;
+ if (pLatencyMs != NULL) {
+ *pLatencyMs = latency;
+ }
return output;
}
@@ -469,13 +479,21 @@ public:
remote()->transact(OPEN_INPUT, data, &reply);
audio_io_handle_t input = (audio_io_handle_t) reply.readInt32();
devices = (audio_devices_t)reply.readInt32();
- if (pDevices != NULL) *pDevices = devices;
+ if (pDevices != NULL) {
+ *pDevices = devices;
+ }
samplingRate = reply.readInt32();
- if (pSamplingRate != NULL) *pSamplingRate = samplingRate;
+ if (pSamplingRate != NULL) {
+ *pSamplingRate = samplingRate;
+ }
format = (audio_format_t) reply.readInt32();
- if (pFormat != NULL) *pFormat = format;
+ if (pFormat != NULL) {
+ *pFormat = format;
+ }
channelMask = (audio_channel_mask_t)reply.readInt32();
- if (pChannelMask != NULL) *pChannelMask = channelMask;
+ if (pChannelMask != NULL) {
+ *pChannelMask = channelMask;
+ }
return input;
}
@@ -517,11 +535,11 @@ public:
status_t status = reply.readInt32();
if (status == NO_ERROR) {
uint32_t tmp = reply.readInt32();
- if (halFrames) {
+ if (halFrames != NULL) {
*halFrames = tmp;
}
tmp = reply.readInt32();
- if (dspFrames) {
+ if (dspFrames != NULL) {
*dspFrames = tmp;
}
}
@@ -639,7 +657,7 @@ public:
if (pDesc == NULL) {
return effect;
- if (status) {
+ if (status != NULL) {
*status = BAD_VALUE;
}
}
@@ -657,7 +675,7 @@ public:
} else {
lStatus = reply.readInt32();
int tmp = reply.readInt32();
- if (id) {
+ if (id != NULL) {
*id = tmp;
}
tmp = reply.readInt32();
@@ -667,7 +685,7 @@ public:
effect = interface_cast<IEffect>(reply.readStrongBinder());
reply.read(pDesc, sizeof(effect_descriptor_t));
}
- if (status) {
+ if (status != NULL) {
*status = lStatus;
}
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 3d65c44..e81e253 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1421,6 +1421,7 @@ audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module,
{
PlaybackThread *thread = NULL;
struct audio_config config;
+ memset(&config, 0, sizeof(config));
config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
@@ -1645,6 +1646,7 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
status_t status;
RecordThread *thread = NULL;
struct audio_config config;
+ memset(&config, 0, sizeof(config));
config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index ffe3e9f..6c0d1d3 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -57,5 +57,4 @@ private:
// releaseBuffer() not overridden
bool mOverflow; // overflow on most recent attempt to fill client buffer
- AudioRecordServerProxy* mAudioRecordServerProxy;
};
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 2c2931f..59d42a5 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -266,8 +266,8 @@ AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio
: Thread(false /*canCallJava*/),
mType(type),
mAudioFlinger(audioFlinger),
- // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, and mFormat are
- // set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters()
+ // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize
+ // are set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters()
mParamStatus(NO_ERROR),
mStandby(false), mOutDevice(outDevice), mInDevice(inDevice),
mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id),
@@ -423,6 +423,8 @@ void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
result.append(buffer);
snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount);
result.append(buffer);
+ snprintf(buffer, SIZE, "HAL buffer size: %u bytes\n", mBufferSize);
+ result.append(buffer);
snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount);
result.append(buffer);
snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
@@ -1559,7 +1561,8 @@ void AudioFlinger::PlaybackThread::readOutputParameters()
mFormat);
}
mFrameSize = audio_stream_frame_size(&mOutput->stream->common);
- mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
+ mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common);
+ mFrameCount = mBufferSize / mFrameSize;
if (mFrameCount & 15) {
ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames",
mFrameCount);
@@ -4159,7 +4162,7 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
) :
ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD),
mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL),
- // mRsmpInIndex and mBufferSize set by readInputParameters()
+ // mRsmpInIndex set by readInputParameters()
mReqChannelCount(popcount(channelMask)),
mReqSampleRate(sampleRate)
// mBytesRead is only meaningful while active, and so is cleared in start()
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 31d5323..3ad9b25 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -270,6 +270,7 @@ protected:
uint32_t mChannelCount;
size_t mFrameSize;
audio_format_t mFormat;
+ size_t mBufferSize; // HAL buffer size for read() or write()
// Parameter sequence by client: binder thread calling setParameters():
// 1. Lock mLock
@@ -903,7 +904,6 @@ private:
int32_t *mRsmpOutBuffer;
int16_t *mRsmpInBuffer; // [mFrameCount * mChannelCount]
size_t mRsmpInIndex;
- size_t mBufferSize; // stream buffer size for read()
const uint32_t mReqChannelCount;
const uint32_t mReqSampleRate;
ssize_t mBytesRead;
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index e676365..6ea6f2f 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -1651,9 +1651,7 @@ AudioFlinger::RecordThread::RecordTrack::RecordTrack(
{
ALOGV("RecordTrack constructor");
if (mCblk != NULL) {
- mAudioRecordServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount,
- mFrameSize);
- mServerProxy = mAudioRecordServerProxy;
+ mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, mFrameSize);
}
}
generated by cgit v1.1 at 2024-09-24 21:33:43 +0000