Merge "Improve dynamic audio resampler filter generation"
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index cd855da..ca9aaf7 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -94,6 +94,8 @@
static float linearToLog(int volume);
static int logToLinear(float volume);
+ // Returned samplingRate and frameCount output values are guaranteed
+ // to be non-zero if status == NO_ERROR
static status_t getOutputSamplingRate(uint32_t* samplingRate,
audio_stream_type_t stream);
static status_t getOutputFrameCount(size_t* frameCount,
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 9f631cf..8afcc5d 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -123,6 +123,8 @@
* - NO_ERROR: successful operation
* - NO_INIT: audio server or audio hardware not initialized
* - BAD_VALUE: unsupported configuration
+ * frameCount is guaranteed to be non-zero if status is NO_ERROR,
+ * and is undefined otherwise.
*/
static status_t getMinFrameCount(size_t* frameCount,
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 9a76111..4580030 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -40,10 +40,10 @@
DefaultKeyedVector<audio_io_handle_t, AudioSystem::OutputDescriptor *> AudioSystem::gOutputs(0);
// Cached values for recording queries, all protected by gLock
-uint32_t AudioSystem::gPrevInSamplingRate = 16000;
-audio_format_t AudioSystem::gPrevInFormat = AUDIO_FORMAT_PCM_16_BIT;
-audio_channel_mask_t AudioSystem::gPrevInChannelMask = AUDIO_CHANNEL_IN_MONO;
-size_t AudioSystem::gInBuffSize = 0;
+uint32_t AudioSystem::gPrevInSamplingRate;
+audio_format_t AudioSystem::gPrevInFormat;
+audio_channel_mask_t AudioSystem::gPrevInChannelMask;
+size_t AudioSystem::gInBuffSize = 0; // zero indicates cache is invalid
// establish binder interface to AudioFlinger service
@@ -259,6 +259,11 @@
*samplingRate = outputDesc->samplingRate;
gLock.unlock();
}
+ if (*samplingRate == 0) {
+ ALOGE("AudioSystem::getSamplingRate failed for output %d stream type %d",
+ output, streamType);
+ return BAD_VALUE;
+ }
ALOGV("getSamplingRate() streamType %d, output %d, sampling rate %u", streamType, output,
*samplingRate);
@@ -299,6 +304,11 @@
*frameCount = outputDesc->frameCount;
gLock.unlock();
}
+ if (*frameCount == 0) {
+ ALOGE("AudioSystem::getFrameCount failed for output %d stream type %d",
+ output, streamType);
+ return BAD_VALUE;
+ }
ALOGV("getFrameCount() streamType %d, output %d, frameCount %d", streamType, output,
*frameCount);
@@ -359,6 +369,12 @@
return PERMISSION_DENIED;
}
inBuffSize = af->getInputBufferSize(sampleRate, format, channelMask);
+ if (inBuffSize == 0) {
+ ALOGE("AudioSystem::getInputBufferSize failed sampleRate %d format %x channelMask %x",
+ sampleRate, format, channelMask);
+ return BAD_VALUE;
+ }
+ // A benign race is possible here: we could overwrite a fresher cache entry
gLock.lock();
// save the request params
gPrevInSamplingRate = sampleRate;
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index ddd4612..a6ffc62 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -44,9 +44,6 @@
return BAD_VALUE;
}
- // default to 0 in case of error
- *frameCount = 0;
-
// FIXME merge with similar code in createTrack_l(), except we're missing
// some information here that is available in createTrack_l():
// audio_io_handle_t output
@@ -54,16 +51,20 @@
// audio_channel_mask_t channelMask
// audio_output_flags_t flags
uint32_t afSampleRate;
- if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
- return NO_INIT;
+ status_t status;
+ status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType);
+ if (status != NO_ERROR) {
+ return status;
}
size_t afFrameCount;
- if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
- return NO_INIT;
+ status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType);
+ if (status != NO_ERROR) {
+ return status;
}
uint32_t afLatency;
- if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
- return NO_INIT;
+ status = AudioSystem::getOutputLatency(&afLatency, streamType);
+ if (status != NO_ERROR) {
+ return status;
}
// Ensure that buffer depth covers at least audio hardware latency
@@ -74,6 +75,13 @@
*frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
afFrameCount * minBufCount * sampleRate / afSampleRate;
+ // The formula above should always produce a non-zero value, but return an error
+ // in the unlikely event that it does not, as that's part of the API contract.
+ if (*frameCount == 0) {
+ ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %d",
+ streamType, sampleRate);
+ return BAD_VALUE;
+ }
ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d",
*frameCount, afFrameCount, minBufCount, afSampleRate, afLatency);
return NO_ERROR;
diff --git a/media/libstagefright/httplive/M3UParser.cpp b/media/libstagefright/httplive/M3UParser.cpp
index 243888c..dd248cb 100644
--- a/media/libstagefright/httplive/M3UParser.cpp
+++ b/media/libstagefright/httplive/M3UParser.cpp
@@ -608,7 +608,7 @@
if (meta->get() == NULL) {
*meta = new AMessage;
}
- (*meta)->setInt64(key, (int64_t)x * 1E6);
+ (*meta)->setInt64(key, (int64_t)(x * 1E6));
return OK;
}
diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp
index c5ad2c0..c8f0730 100644
--- a/services/audioflinger/AudioPolicyService.cpp
+++ b/services/audioflinger/AudioPolicyService.cpp
@@ -1441,6 +1441,14 @@
loadEffects(root, effects);
loadInputSources(root, effects);
+ // delete effects to fix memory leak.
+ // as effects is local var and valgrind would treat this as memory leak
+ // and although it only did in mediaserver init, but free it in case mediaserver reboot
+ size_t i;
+ for (i = 0; i < effects.size(); i++) {
+ delete effects[i];
+ }
+
config_free(root);
free(root);
free(data);