Merge "Update AudioParameter"
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 7d5d772..6de6486 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -139,13 +139,15 @@
* latency of the track. The actual size selected by the AudioTrack could be
* larger if the requested size is not compatible with current audio HAL
* latency.
- * flags: Reserved for future use.
+ * flags: See comments on audio_policy_output_flags_t in <system/audio_policy.h>.
* cbf: Callback function. If not null, this function is called periodically
* to request new PCM data.
* user: Context for use by the callback receiver.
* notificationFrames: The callback function is called each time notificationFrames PCM
* frames have been consumed from track input buffer.
* sessionId: Specific session ID, or zero to use default.
+ * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI.
+ * If not present in parameter list, then fixed at false.
*/
AudioTrack( audio_stream_type_t streamType,
@@ -157,7 +159,7 @@
callback_t cbf = NULL,
void* user = NULL,
int notificationFrames = 0,
- int sessionId = 0);
+ int sessionId = 0);
// DEPRECATED
explicit AudioTrack( int streamType,
@@ -189,7 +191,7 @@
callback_t cbf = NULL,
void* user = NULL,
int notificationFrames = 0,
- int sessionId = 0);
+ int sessionId = 0);
/* Terminates the AudioTrack and unregisters it from AudioFlinger.
* Also destroys all resources associated with the AudioTrack.
diff --git a/include/media/MediaPlayerInterface.h b/include/media/MediaPlayerInterface.h
index d4aa233..9a8f4b0 100644
--- a/include/media/MediaPlayerInterface.h
+++ b/include/media/MediaPlayerInterface.h
@@ -87,8 +87,9 @@
virtual ssize_t frameSize() const = 0;
virtual uint32_t latency() const = 0;
virtual float msecsPerFrame() const = 0;
- virtual status_t getPosition(uint32_t *position) = 0;
- virtual int getSessionId() = 0;
+ virtual status_t getPosition(uint32_t *position) const = 0;
+ virtual status_t getFramesWritten(uint32_t *frameswritten) const = 0;
+ virtual int getSessionId() const = 0;
// If no callback is specified, use the "write" API below to submit
// audio data.
diff --git a/libvideoeditor/lvpp/VideoEditorPlayer.cpp b/libvideoeditor/lvpp/VideoEditorPlayer.cpp
index 57cab08..1ba1f44 100755
--- a/libvideoeditor/lvpp/VideoEditorPlayer.cpp
+++ b/libvideoeditor/lvpp/VideoEditorPlayer.cpp
@@ -376,12 +376,19 @@
return mMsecsPerFrame;
}
-status_t VideoEditorPlayer::VeAudioOutput::getPosition(uint32_t *position) {
+status_t VideoEditorPlayer::VeAudioOutput::getPosition(uint32_t *position) const {
if (mTrack == 0) return NO_INIT;
return mTrack->getPosition(position);
}
+status_t VideoEditorPlayer::VeAudioOutput::getFramesWritten(uint32_t *written) const {
+
+ if (mTrack == 0) return NO_INIT;
+ *written = mNumFramesWritten;
+ return OK;
+}
+
status_t VideoEditorPlayer::VeAudioOutput::open(
uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
audio_format_t format, int bufferCount,
@@ -569,7 +576,7 @@
return NO_ERROR;
}
-int VideoEditorPlayer::VeAudioOutput::getSessionId() {
+int VideoEditorPlayer::VeAudioOutput::getSessionId() const {
return mSessionId;
}
diff --git a/libvideoeditor/lvpp/VideoEditorPlayer.h b/libvideoeditor/lvpp/VideoEditorPlayer.h
index 6962501..350b384 100755
--- a/libvideoeditor/lvpp/VideoEditorPlayer.h
+++ b/libvideoeditor/lvpp/VideoEditorPlayer.h
@@ -45,8 +45,9 @@
virtual ssize_t frameSize() const;
virtual uint32_t latency() const;
virtual float msecsPerFrame() const;
- virtual status_t getPosition(uint32_t *position);
- virtual int getSessionId();
+ virtual status_t getPosition(uint32_t *position) const;
+ virtual status_t getFramesWritten(uint32_t*) const;
+ virtual int getSessionId() const;
virtual status_t open(
uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index d73eabd..c619ad7 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -226,7 +226,8 @@
// force direct flag if format is not linear PCM
if (!audio_is_linear_pcm(format)) {
- flags = (audio_policy_output_flags_t) (flags | AUDIO_POLICY_OUTPUT_FLAG_DIRECT);
+ flags = (audio_policy_output_flags_t)
+ ((flags | AUDIO_POLICY_OUTPUT_FLAG_DIRECT) & ~AUDIO_POLICY_OUTPUT_FLAG_FAST);
}
if (!audio_is_output_channel(channelMask)) {
@@ -252,6 +253,7 @@
mNotificationFramesReq = notificationFrames;
mSessionId = sessionId;
mAuxEffectId = 0;
+ mCbf = cbf;
// create the IAudioTrack
status_t status = createTrack_l(streamType,
@@ -280,7 +282,6 @@
mSharedBuffer = sharedBuffer;
mMuted = false;
mActive = false;
- mCbf = cbf;
mUserData = user;
mLoopCount = 0;
mMarkerPosition = 0;
@@ -762,6 +763,18 @@
return NO_INIT;
}
+ // Client decides whether the track is TIMED (see below), but can only express a preference
+ // for FAST. Server will perform additional tests.
+ if ((flags & AUDIO_POLICY_OUTPUT_FLAG_FAST) && !(
+ // either of these use cases:
+ // use case 1: shared buffer
+ (sharedBuffer != 0) ||
+ // use case 2: callback handler
+ (mCbf != NULL))) {
+ ALOGW("AUDIO_POLICY_OUTPUT_FLAG_FAST denied");
+ flags = (audio_policy_output_flags_t) (flags & ~AUDIO_POLICY_OUTPUT_FLAG_FAST);
+ }
+
mNotificationFramesAct = mNotificationFramesReq;
if (!audio_is_linear_pcm(format)) {
if (sharedBuffer != 0) {
@@ -786,7 +799,7 @@
if (mNotificationFramesAct > (uint32_t)frameCount/2) {
mNotificationFramesAct = frameCount/2;
}
- if (frameCount < minFrameCount) {
+ if (frameCount < minFrameCount && !(flags & AUDIO_POLICY_OUTPUT_FLAG_FAST)) {
// not ALOGW because it happens all the time when playing key clicks over A2DP
ALOGV("Minimum buffer size corrected from %d to %d",
frameCount, minFrameCount);
@@ -807,6 +820,10 @@
if (mIsTimed) {
trackFlags |= IAudioFlinger::TRACK_TIMED;
}
+ if (flags & AUDIO_POLICY_OUTPUT_FLAG_FAST) {
+ trackFlags |= IAudioFlinger::TRACK_FAST;
+ }
+
sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
streamType,
sampleRate,
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index cc3138d..a977337 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -1417,6 +1417,7 @@
: mCallback(NULL),
mCallbackCookie(NULL),
mCallbackData(NULL),
+ mBytesWritten(0),
mSessionId(sessionId) {
ALOGV("AudioOutput(%d)", sessionId);
mTrack = 0;
@@ -1495,12 +1496,19 @@
return mMsecsPerFrame;
}
-status_t MediaPlayerService::AudioOutput::getPosition(uint32_t *position)
+status_t MediaPlayerService::AudioOutput::getPosition(uint32_t *position) const
{
if (mTrack == 0) return NO_INIT;
return mTrack->getPosition(position);
}
+status_t MediaPlayerService::AudioOutput::getFramesWritten(uint32_t *frameswritten) const
+{
+ if (mTrack == 0) return NO_INIT;
+ *frameswritten = mBytesWritten / frameSize();
+ return OK;
+}
+
status_t MediaPlayerService::AudioOutput::open(
uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
audio_format_t format, int bufferCount,
@@ -1656,6 +1664,7 @@
mTrack = NULL;
mNextOutput->mSampleRateHz = mSampleRateHz;
mNextOutput->mMsecsPerFrame = mMsecsPerFrame;
+ mNextOutput->mBytesWritten = mBytesWritten;
}
}
@@ -1666,6 +1675,7 @@
//ALOGV("write(%p, %u)", buffer, size);
if (mTrack) {
ssize_t ret = mTrack->write(buffer, size);
+ mBytesWritten += ret;
return ret;
}
return NO_INIT;
@@ -1777,7 +1787,7 @@
data->unlock();
}
-int MediaPlayerService::AudioOutput::getSessionId()
+int MediaPlayerService::AudioOutput::getSessionId() const
{
return mSessionId;
}
@@ -1802,13 +1812,20 @@
return mMsecsPerFrame;
}
-status_t MediaPlayerService::AudioCache::getPosition(uint32_t *position)
+status_t MediaPlayerService::AudioCache::getPosition(uint32_t *position) const
{
if (position == 0) return BAD_VALUE;
*position = mSize;
return NO_ERROR;
}
+status_t MediaPlayerService::AudioCache::getFramesWritten(uint32_t *written) const
+{
+ if (written == 0) return BAD_VALUE;
+ *written = mSize;
+ return NO_ERROR;
+}
+
////////////////////////////////////////////////////////////////////////////////
struct CallbackThread : public Thread {
@@ -1971,7 +1988,7 @@
p->mSignal.signal();
}
-int MediaPlayerService::AudioCache::getSessionId()
+int MediaPlayerService::AudioCache::getSessionId() const
{
return 0;
}
diff --git a/media/libmediaplayerservice/MediaPlayerService.h b/media/libmediaplayerservice/MediaPlayerService.h
index b08dd6c..2a8cfd2 100644
--- a/media/libmediaplayerservice/MediaPlayerService.h
+++ b/media/libmediaplayerservice/MediaPlayerService.h
@@ -84,8 +84,9 @@
virtual ssize_t frameSize() const;
virtual uint32_t latency() const;
virtual float msecsPerFrame() const;
- virtual status_t getPosition(uint32_t *position);
- virtual int getSessionId();
+ virtual status_t getPosition(uint32_t *position) const;
+ virtual status_t getFramesWritten(uint32_t *frameswritten) const;
+ virtual int getSessionId() const;
virtual status_t open(
uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
@@ -122,6 +123,7 @@
AudioCallback mCallback;
void * mCallbackCookie;
CallbackData * mCallbackData;
+ uint64_t mBytesWritten;
audio_stream_type_t mStreamType;
float mLeftVolume;
float mRightVolume;
@@ -181,8 +183,9 @@
virtual ssize_t frameSize() const { return ssize_t(mChannelCount * ((mFormat == AUDIO_FORMAT_PCM_16_BIT)?sizeof(int16_t):sizeof(u_int8_t))); }
virtual uint32_t latency() const;
virtual float msecsPerFrame() const;
- virtual status_t getPosition(uint32_t *position);
- virtual int getSessionId();
+ virtual status_t getPosition(uint32_t *position) const;
+ virtual status_t getFramesWritten(uint32_t *frameswritten) const;
+ virtual int getSessionId() const;
virtual status_t open(
uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
index 544d501..11cea3b 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
@@ -39,6 +39,7 @@
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
+#include <media/stagefright/SkipCutBuffer.h>
#include <gui/ISurfaceTexture.h>
#include "avc_utils.h"
@@ -63,10 +64,13 @@
mSkipRenderingVideoUntilMediaTimeUs(-1ll),
mVideoLateByUs(0ll),
mNumFramesTotal(0ll),
- mNumFramesDropped(0ll) {
+ mNumFramesDropped(0ll),
+ mSkipCutBuffer(NULL) {
}
NuPlayer::~NuPlayer() {
+ delete mSkipCutBuffer;
+ mSkipCutBuffer = NULL;
}
void NuPlayer::setUID(uid_t uid) {
@@ -234,6 +238,32 @@
mSource->start();
+ sp<MetaData> meta = mSource->getFormat(true /* audio */);
+ if (meta != NULL) {
+ int32_t delay = 0;
+ if (!meta->findInt32(kKeyEncoderDelay, &delay)) {
+ delay = 0;
+ }
+ int32_t padding = 0;
+ if (!meta->findInt32(kKeyEncoderPadding, &padding)) {
+ padding = 0;
+ }
+ int32_t numchannels = 0;
+ if (delay + padding) {
+ if (meta->findInt32(kKeyChannelCount, &numchannels)) {
+ size_t frameSize = numchannels * sizeof(int16_t);
+ if (mSkipCutBuffer) {
+ size_t prevbuffersize = mSkipCutBuffer->size();
+ if (prevbuffersize != 0) {
+ ALOGW("Replacing SkipCutBuffer holding %d bytes", prevbuffersize);
+ }
+ delete mSkipCutBuffer;
+ }
+ mSkipCutBuffer = new SkipCutBuffer(delay * frameSize, padding * frameSize);
+ }
+ }
+ }
+
mRenderer = new Renderer(
mAudioSink,
new AMessage(kWhatRendererNotify, id()));
@@ -844,6 +874,10 @@
skipUntilMediaTimeUs = -1;
}
+ if (audio && mSkipCutBuffer) {
+ mSkipCutBuffer->submit(buffer);
+ }
+
mRenderer->queueBuffer(audio, buffer, reply);
}
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.h b/media/libmediaplayerservice/nuplayer/NuPlayer.h
index 25766e0..f917f64 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.h
@@ -27,6 +27,7 @@
struct ACodec;
struct MetaData;
struct NuPlayerDriver;
+class SkipCutBuffer;
struct NuPlayer : public AHandler {
NuPlayer();
@@ -128,6 +129,8 @@
int64_t mVideoLateByUs;
int64_t mNumFramesTotal, mNumFramesDropped;
+ SkipCutBuffer *mSkipCutBuffer;
+
status_t instantiateDecoder(bool audio, sp<Decoder> *decoder);
status_t feedDecoderInputData(bool audio, const sp<AMessage> &msg);
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
index ecbc428..1f13955 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
@@ -591,6 +591,10 @@
void NuPlayer::Renderer::onAudioSinkChanged() {
CHECK(!mDrainAudioQueuePending);
mNumFramesWritten = 0;
+ uint32_t written;
+ if (mAudioSink->getFramesWritten(&written) == OK) {
+ mNumFramesWritten = written;
+ }
}
void NuPlayer::Renderer::notifyPosition() {
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 83af5f3..fb5a7e1 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1924,7 +1924,7 @@
}
// this method must always be called either with ThreadBase mLock held or inside the thread loop
-audio_stream_t* AudioFlinger::PlaybackThread::stream()
+audio_stream_t* AudioFlinger::PlaybackThread::stream() const
{
if (mOutput == NULL) {
return NULL;
@@ -1932,7 +1932,7 @@
return &mOutput->stream->common;
}
-uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs()
+uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const
{
// A2DP output latency is not due only to buffering capacity. It also reflects encoding,
// decoding and transfer time. So sleeping for half of the latency would likely cause
@@ -2817,12 +2817,12 @@
return NO_ERROR;
}
-uint32_t AudioFlinger::MixerThread::idleSleepTimeUs()
+uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const
{
return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
}
-uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs()
+uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const
{
return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
}
@@ -3166,7 +3166,7 @@
return reconfig;
}
-uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs()
+uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const
{
uint32_t time;
if (audio_is_linear_pcm(mFormat)) {
@@ -3177,7 +3177,7 @@
return time;
}
-uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs()
+uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const
{
uint32_t time;
if (audio_is_linear_pcm(mFormat)) {
@@ -3188,7 +3188,7 @@
return time;
}
-uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs()
+uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const
{
uint32_t time;
if (audio_is_linear_pcm(mFormat)) {
@@ -3351,7 +3351,7 @@
return true;
}
-uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs()
+uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const
{
return (mWaitTimeMs * 1000) / 2;
}
@@ -5623,7 +5623,7 @@
}
// this method must always be called either with ThreadBase mLock held or inside the thread loop
-audio_stream_t* AudioFlinger::RecordThread::stream()
+audio_stream_t* AudioFlinger::RecordThread::stream() const
{
if (mInput == NULL) {
return NULL;
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 2376aff..3051514 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -472,7 +472,7 @@
audio_io_handle_t id() const { return mId;}
bool standby() const { return mStandby; }
uint32_t device() const { return mDevice; }
- virtual audio_stream_t* stream() = 0;
+ virtual audio_stream_t* stream() const = 0;
sp<EffectHandle> createEffect_l(
const sp<AudioFlinger::Client>& client,
@@ -918,7 +918,7 @@
AudioStreamOut* getOutput() const;
AudioStreamOut* clearOutput();
- virtual audio_stream_t* stream();
+ virtual audio_stream_t* stream() const;
void suspend() { mSuspended++; }
void restore() { if (mSuspended > 0) mSuspended--; }
@@ -960,9 +960,13 @@
// Allocate a track name. Returns name >= 0 if successful, -1 on failure.
virtual int getTrackName_l() = 0;
virtual void deleteTrackName_l(int name) = 0;
- virtual uint32_t activeSleepTimeUs();
- virtual uint32_t idleSleepTimeUs() = 0;
- virtual uint32_t suspendSleepTimeUs() = 0;
+
+ // Time to sleep between cycles when:
+ virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED
+ virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE
+ virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us
+ // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write()
+ // No sleep in standby mode; waits on a condition
// Code snippets that are temporarily lifted up out of threadLoop() until the merge
void checkSilentMode_l();
@@ -1048,8 +1052,8 @@
virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
virtual int getTrackName_l();
virtual void deleteTrackName_l(int name);
- virtual uint32_t idleSleepTimeUs();
- virtual uint32_t suspendSleepTimeUs();
+ virtual uint32_t idleSleepTimeUs() const;
+ virtual uint32_t suspendSleepTimeUs() const;
virtual void cacheParameters_l();
// threadLoop snippets
@@ -1073,9 +1077,9 @@
protected:
virtual int getTrackName_l();
virtual void deleteTrackName_l(int name);
- virtual uint32_t activeSleepTimeUs();
- virtual uint32_t idleSleepTimeUs();
- virtual uint32_t suspendSleepTimeUs();
+ virtual uint32_t activeSleepTimeUs() const;
+ virtual uint32_t idleSleepTimeUs() const;
+ virtual uint32_t suspendSleepTimeUs() const;
virtual void cacheParameters_l();
// threadLoop snippets
@@ -1110,9 +1114,9 @@
// Thread virtuals
void addOutputTrack(MixerThread* thread);
void removeOutputTrack(MixerThread* thread);
- uint32_t waitTimeMs() { return mWaitTimeMs; }
+ uint32_t waitTimeMs() const { return mWaitTimeMs; }
protected:
- virtual uint32_t activeSleepTimeUs();
+ virtual uint32_t activeSleepTimeUs() const;
private:
bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks);
@@ -1260,7 +1264,7 @@
status_t dump(int fd, const Vector<String16>& args);
AudioStreamIn* getInput() const;
AudioStreamIn* clearInput();
- virtual audio_stream_t* stream();
+ virtual audio_stream_t* stream() const;
// AudioBufferProvider interface
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts);
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 3f4c19a..0e6ea12 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -66,32 +66,7 @@
// and mTrackNames is initially 0. However, leave it here until that's verified.
track_t* t = mState.tracks;
for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {
- t->needs = 0;
- t->volume[0] = UNITY_GAIN;
- t->volume[1] = UNITY_GAIN;
- // no initialization needed
- // t->prevVolume[0]
- // t->prevVolume[1]
- t->volumeInc[0] = 0;
- t->volumeInc[1] = 0;
- t->auxLevel = 0;
- t->auxInc = 0;
- // no initialization needed
- // t->prevAuxLevel
- // t->frameCount
- t->channelCount = 2;
- t->enabled = false;
- t->format = 16;
- t->channelMask = AUDIO_CHANNEL_OUT_STEREO;
- t->bufferProvider = NULL;
- t->buffer.raw = NULL;
- // t->buffer.frameCount
- t->hook = NULL;
- t->in = NULL;
- t->resampler = NULL;
- t->sampleRate = mSampleRate;
- t->mainBuffer = NULL;
- t->auxBuffer = NULL;
+ // FIXME redundant per track
t->localTimeFreq = lc.getLocalFreq();
t++;
}
@@ -115,6 +90,38 @@
int n = __builtin_ctz(names);
ALOGV("add track (%d)", n);
mTrackNames |= 1 << n;
+ // assume default parameters for the track, except where noted below
+ track_t* t = &mState.tracks[n];
+ t->needs = 0;
+ t->volume[0] = UNITY_GAIN;
+ t->volume[1] = UNITY_GAIN;
+ // no initialization needed
+ // t->prevVolume[0]
+ // t->prevVolume[1]
+ t->volumeInc[0] = 0;
+ t->volumeInc[1] = 0;
+ t->auxLevel = 0;
+ t->auxInc = 0;
+ // no initialization needed
+ // t->prevAuxLevel
+ // t->frameCount
+ t->channelCount = 2;
+ t->enabled = false;
+ t->format = 16;
+ t->channelMask = AUDIO_CHANNEL_OUT_STEREO;
+ // setBufferProvider(name, AudioBufferProvider *) is required before enable(name)
+ t->bufferProvider = NULL;
+ t->buffer.raw = NULL;
+ // no initialization needed
+ // t->buffer.frameCount
+ t->hook = NULL;
+ t->in = NULL;
+ t->resampler = NULL;
+ t->sampleRate = mSampleRate;
+ // setParameter(name, TRACK, MAIN_BUFFER, mixBuffer) is required before enable(name)
+ t->mainBuffer = NULL;
+ t->auxBuffer = NULL;
+ // see t->localTimeFreq in constructor above
return TRACK0 + n;
}
return -1;
@@ -215,6 +222,9 @@
invalidateState(1 << name);
}
break;
+ case FORMAT:
+ ALOG_ASSERT(valueInt == AUDIO_FORMAT_PCM_16_BIT);
+ break;
default:
LOG_FATAL("bad param");
}