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-rw-r--r--include/media/AudioTrack.h12
-rw-r--r--include/media/stagefright/FileSource.h1
-rwxr-xr-xlibvideoeditor/lvpp/VideoEditorAudioPlayer.cpp3
-rw-r--r--media/libmedia/AudioSystem.cpp2
-rw-r--r--media/libmedia/AudioTrackShared.cpp6
-rw-r--r--media/libmedia/IAudioFlinger.cpp2
-rw-r--r--media/libmedia/IAudioPolicyService.cpp1
-rw-r--r--media/libmediaplayerservice/MediaPlayerService.cpp2
-rw-r--r--media/libstagefright/AudioPlayer.cpp3
-rw-r--r--services/audioflinger/AudioMixer.cpp8
-rw-r--r--services/audioflinger/FastMixer.cpp2
-rw-r--r--services/audioflinger/Threads.cpp18
-rw-r--r--services/audioflinger/Tracks.cpp2
13 files changed, 30 insertions, 32 deletions
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 3dadf52..2c48bbf 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -77,6 +77,7 @@ public:
// (currently ignored, but will make the primary field in future)
size_t size; // input/output in bytes == frameCount * frameSize
+ // on input it is unused
// on output is the number of bytes actually filled
// FIXME this is redundant with respect to frameCount,
// and TRANSFER_OBTAIN mode is broken for 8-bit data
@@ -86,7 +87,7 @@ public:
void* raw;
short* i16; // signed 16-bit
int8_t* i8; // unsigned 8-bit, offset by 0x80
- };
+ }; // input: unused, output: pointer to buffer
};
/* As a convenience, if a callback is supplied, a handler thread
@@ -529,15 +530,6 @@ private:
struct timespec *elapsed = NULL, size_t *nonContig = NULL);
public:
-//EL_FIXME to be reconciled with new obtainBuffer() return codes and control block proxy
-// enum {
-// NO_MORE_BUFFERS = 0x80000001, // same name in AudioFlinger.h, ok to be different value
-// TEAR_DOWN = 0x80000002,
-// STOPPED = 1,
-// STREAM_END_WAIT,
-// STREAM_END
-// };
-
/* Release a filled buffer of "audioBuffer->frameCount" frames for AudioFlinger to process. */
// FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed
void releaseBuffer(Buffer* audioBuffer);
diff --git a/include/media/stagefright/FileSource.h b/include/media/stagefright/FileSource.h
index d994cb3..9838ed2 100644
--- a/include/media/stagefright/FileSource.h
+++ b/include/media/stagefright/FileSource.h
@@ -30,6 +30,7 @@ namespace android {
class FileSource : public DataSource {
public:
FileSource(const char *filename);
+ // FileSource takes ownership and will close the fd
FileSource(int fd, int64_t offset, int64_t length);
virtual status_t initCheck() const;
diff --git a/libvideoeditor/lvpp/VideoEditorAudioPlayer.cpp b/libvideoeditor/lvpp/VideoEditorAudioPlayer.cpp
index cb4b23e..e60030e 100755
--- a/libvideoeditor/lvpp/VideoEditorAudioPlayer.cpp
+++ b/libvideoeditor/lvpp/VideoEditorAudioPlayer.cpp
@@ -534,7 +534,8 @@ status_t VideoEditorAudioPlayer::start(bool sourceAlreadyStarted) {
mAudioTrack = new AudioTrack(
AUDIO_STREAM_MUSIC, mSampleRate, AUDIO_FORMAT_PCM_16_BIT,
audio_channel_out_mask_from_count(numChannels),
- 0, AUDIO_OUTPUT_FLAG_NONE, &AudioCallback, this, 0);
+ 0 /*frameCount*/, AUDIO_OUTPUT_FLAG_NONE, &AudioCallback, this,
+ 0 /*notificationFrames*/);
if ((err = mAudioTrack->initCheck()) != OK) {
mAudioTrack.clear();
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 301c7d6..2f16444 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -35,8 +35,8 @@ Mutex AudioSystem::gLock;
sp<IAudioFlinger> AudioSystem::gAudioFlinger;
sp<AudioSystem::AudioFlingerClient> AudioSystem::gAudioFlingerClient;
audio_error_callback AudioSystem::gAudioErrorCallback = NULL;
-// Cached values
+// Cached values for output handles
DefaultKeyedVector<audio_io_handle_t, AudioSystem::OutputDescriptor *> AudioSystem::gOutputs(NULL);
// Cached values for recording queries, all protected by gLock
diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp
index fdd1a12..58c9fc1 100644
--- a/media/libmedia/AudioTrackShared.cpp
+++ b/media/libmedia/AudioTrackShared.cpp
@@ -200,7 +200,7 @@ status_t ClientProxy::obtainBuffer(Buffer* buffer, const struct timespec *reques
ts = &remaining;
break;
default:
- LOG_FATAL("obtainBuffer() timeout=%d", timeout);
+ LOG_ALWAYS_FATAL("obtainBuffer() timeout=%d", timeout);
ts = NULL;
break;
}
@@ -429,7 +429,7 @@ status_t AudioTrackClientProxy::waitStreamEndDone(const struct timespec *request
ts = &remaining;
break;
default:
- LOG_FATAL("waitStreamEndDone() timeout=%d", timeout);
+ LOG_ALWAYS_FATAL("waitStreamEndDone() timeout=%d", timeout);
ts = NULL;
break;
}
@@ -470,7 +470,7 @@ StaticAudioTrackClientProxy::StaticAudioTrackClientProxy(audio_track_cblk_t* cbl
void StaticAudioTrackClientProxy::flush()
{
- LOG_FATAL("static flush");
+ LOG_ALWAYS_FATAL("static flush");
}
void StaticAudioTrackClientProxy::setLoop(size_t loopStart, size_t loopEnd, int loopCount)
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index 2cb0c5c..eb813bd 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -109,6 +109,7 @@ public:
data.writeInt32(frameCount);
track_flags_t lFlags = flags != NULL ? *flags : (track_flags_t) TRACK_DEFAULT;
data.writeInt32(lFlags);
+ // haveSharedBuffer
if (sharedBuffer != 0) {
data.writeInt32(true);
data.writeStrongBinder(sharedBuffer->asBinder());
@@ -424,6 +425,7 @@ public:
data.writeInt32(channelMask);
data.writeInt32(latency);
data.writeInt32((int32_t) flags);
+ // hasOffloadInfo
if (offloadInfo == NULL) {
data.writeInt32(0);
} else {
diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
index 1a027a6..9bb4a49 100644
--- a/media/libmedia/IAudioPolicyService.cpp
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -137,6 +137,7 @@ public:
data.writeInt32(static_cast <uint32_t>(format));
data.writeInt32(channelMask);
data.writeInt32(static_cast <uint32_t>(flags));
+ // hasOffloadInfo
if (offloadInfo == NULL) {
data.writeInt32(0);
} else {
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index 200c561..778eb9a 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -1679,7 +1679,7 @@ void MediaPlayerService::AudioOutput::switchToNextOutput() {
ssize_t MediaPlayerService::AudioOutput::write(const void* buffer, size_t size)
{
- LOG_FATAL_IF(mCallback != NULL, "Don't call write if supplying a callback.");
+ LOG_ALWAYS_FATAL_IF(mCallback != NULL, "Don't call write if supplying a callback.");
//ALOGV("write(%p, %u)", buffer, size);
if (mTrack != 0) {
diff --git a/media/libstagefright/AudioPlayer.cpp b/media/libstagefright/AudioPlayer.cpp
index 8623100..2669849 100644
--- a/media/libstagefright/AudioPlayer.cpp
+++ b/media/libstagefright/AudioPlayer.cpp
@@ -221,7 +221,8 @@ status_t AudioPlayer::start(bool sourceAlreadyStarted) {
mAudioTrack = new AudioTrack(
AUDIO_STREAM_MUSIC, mSampleRate, AUDIO_FORMAT_PCM_16_BIT, audioMask,
- 0, AUDIO_OUTPUT_FLAG_NONE, &AudioCallback, this, 0);
+ 0 /*frameCount*/, AUDIO_OUTPUT_FLAG_NONE, &AudioCallback, this,
+ 0 /*notificationFrames*/);
if ((err = mAudioTrack->initCheck()) != OK) {
mAudioTrack.clear();
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 3ac5da9..a1783fe 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -449,7 +449,7 @@ void AudioMixer::setParameter(int name, int target, int param, void *value)
}
} break;
default:
- LOG_FATAL("bad param");
+ LOG_ALWAYS_FATAL("setParameter track: bad param %d", param);
}
break;
@@ -474,7 +474,7 @@ void AudioMixer::setParameter(int name, int target, int param, void *value)
invalidateState(1 << name);
break;
default:
- LOG_FATAL("bad param");
+ LOG_ALWAYS_FATAL("setParameter resample: bad param %d", param);
}
break;
@@ -522,12 +522,12 @@ void AudioMixer::setParameter(int name, int target, int param, void *value)
}
break;
default:
- LOG_FATAL("bad param");
+ LOG_ALWAYS_FATAL("setParameter volume: bad param %d", param);
}
break;
default:
- LOG_FATAL("bad target");
+ LOG_ALWAYS_FATAL("setParameter: bad target %d", target);
}
}
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index 90122e0..adb4aca 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -212,7 +212,7 @@ bool FastMixer::threadLoop()
case FastMixerState::MIX_WRITE:
break;
default:
- LOG_FATAL("bad command %d", command);
+ LOG_ALWAYS_FATAL("bad command %d", command);
}
// there is a non-idle state available to us; did the state change?
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index c3c9033..7700780 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -1711,24 +1711,24 @@ int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event,
void AudioFlinger::PlaybackThread::readOutputParameters_l()
{
- // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL
+ // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL
mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
if (!audio_is_output_channel(mChannelMask)) {
- LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask);
+ LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask);
}
if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) {
- LOG_FATAL("HAL channel mask %#x not supported for mixed output; "
+ LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output; "
"must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask);
}
mChannelCount = popcount(mChannelMask);
mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
if (!audio_is_valid_format(mFormat)) {
- LOG_FATAL("HAL format %#x not valid for output", mFormat);
+ LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat);
}
if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) {
- LOG_FATAL("HAL format %#x not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT",
- mFormat);
+ LOG_ALWAYS_FATAL("HAL format %#x not supported for mixed output; "
+ "must be AUDIO_FORMAT_PCM_16_BIT", mFormat);
}
mFrameSize = audio_stream_frame_size(&mOutput->stream->common);
mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common);
@@ -3118,7 +3118,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
break;
case TrackBase::IDLE:
default:
- LOG_FATAL("unexpected track state %d", track->mState);
+ LOG_ALWAYS_FATAL("unexpected track state %d", track->mState);
}
if (isActive) {
@@ -3149,7 +3149,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
// because we're about to decrement the last sp<> on those tracks.
block = FastMixerStateQueue::BLOCK_UNTIL_ACKED;
} else {
- LOG_FATAL("fast track %d should have been active", j);
+ LOG_ALWAYS_FATAL("fast track %d should have been active", j);
}
tracksToRemove->add(track);
// Avoids a misleading display in dumpsys
@@ -4765,7 +4765,7 @@ reacquire_wakelock:
continue;
default:
- LOG_FATAL("Unexpected activeTrackState %d", activeTrackState);
+ LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState);
}
activeTracks.add(activeTrack);
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 9fe459b..2cf10e2 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -1525,7 +1525,7 @@ void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer(
mTrimQueueHeadOnRelease = false;
}
} else {
- LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no"
+ LOG_ALWAYS_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no"
" buffers in the timed buffer queue");
}
generated by cgit v1.1 at 2024-05-28 22:49:05 +0000