MediaPlayer2: remove MediaPlayer2Engine
Test: MediaPlayer2 plays
Bug: 63934228
Change-Id: I2459082bf6bbe2384ed94901b1f8f28559345888
diff --git a/media/libmediaplayer2/MediaPlayer2AudioOutput.cpp b/media/libmediaplayer2/MediaPlayer2AudioOutput.cpp
new file mode 100644
index 0000000..a8e1d1f
--- /dev/null
+++ b/media/libmediaplayer2/MediaPlayer2AudioOutput.cpp
@@ -0,0 +1,727 @@
+/*
+**
+** Copyright 2018, The Android Open Source Project
+**
+** Licensed under the Apache License, Version 2.0 (the "License");
+** you may not use this file except in compliance with the License.
+** You may obtain a copy of the License at
+**
+** http://www.apache.org/licenses/LICENSE-2.0
+**
+** Unless required by applicable law or agreed to in writing, software
+** distributed under the License is distributed on an "AS IS" BASIS,
+** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+** See the License for the specific language governing permissions and
+** limitations under the License.
+*/
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "MediaPlayer2AudioOutput"
+#include <mediaplayer2/MediaPlayer2AudioOutput.h>
+
+#include <cutils/properties.h> // for property_get
+#include <utils/Log.h>
+
+#include <media/AudioPolicyHelper.h>
+#include <media/AudioTrack.h>
+#include <media/stagefright/foundation/ADebug.h>
+
+namespace {
+
+const float kMaxRequiredSpeed = 8.0f; // for PCM tracks allow up to 8x speedup.
+
+} // anonymous namespace
+
+namespace android {
+
+// TODO: Find real cause of Audio/Video delay in PV framework and remove this workaround
+/* static */ int MediaPlayer2AudioOutput::mMinBufferCount = 4;
+/* static */ bool MediaPlayer2AudioOutput::mIsOnEmulator = false;
+
+status_t MediaPlayer2AudioOutput::dump(int fd, const Vector<String16>& args) const {
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ result.append(" MediaPlayer2AudioOutput\n");
+ snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n",
+ mStreamType, mLeftVolume, mRightVolume);
+ result.append(buffer);
+ snprintf(buffer, 255, " msec per frame(%f), latency (%d)\n",
+ mMsecsPerFrame, (mTrack != 0) ? mTrack->latency() : -1);
+ result.append(buffer);
+ snprintf(buffer, 255, " aux effect id(%d), send level (%f)\n",
+ mAuxEffectId, mSendLevel);
+ result.append(buffer);
+
+ ::write(fd, result.string(), result.size());
+ if (mTrack != 0) {
+ mTrack->dump(fd, args);
+ }
+ return NO_ERROR;
+}
+
+MediaPlayer2AudioOutput::MediaPlayer2AudioOutput(audio_session_t sessionId, uid_t uid, int pid,
+ const audio_attributes_t* attr, const sp<AudioSystem::AudioDeviceCallback>& deviceCallback)
+ : mCallback(NULL),
+ mCallbackCookie(NULL),
+ mCallbackData(NULL),
+ mStreamType(AUDIO_STREAM_MUSIC),
+ mLeftVolume(1.0),
+ mRightVolume(1.0),
+ mPlaybackRate(AUDIO_PLAYBACK_RATE_DEFAULT),
+ mSampleRateHz(0),
+ mMsecsPerFrame(0),
+ mFrameSize(0),
+ mSessionId(sessionId),
+ mUid(uid),
+ mPid(pid),
+ mSendLevel(0.0),
+ mAuxEffectId(0),
+ mFlags(AUDIO_OUTPUT_FLAG_NONE),
+ mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
+ mRoutedDeviceId(AUDIO_PORT_HANDLE_NONE),
+ mDeviceCallbackEnabled(false),
+ mDeviceCallback(deviceCallback) {
+ ALOGV("MediaPlayer2AudioOutput(%d)", sessionId);
+ if (attr != NULL) {
+ mAttributes = (audio_attributes_t *) calloc(1, sizeof(audio_attributes_t));
+ if (mAttributes != NULL) {
+ memcpy(mAttributes, attr, sizeof(audio_attributes_t));
+ mStreamType = audio_attributes_to_stream_type(attr);
+ }
+ } else {
+ mAttributes = NULL;
+ }
+
+ setMinBufferCount();
+}
+
+MediaPlayer2AudioOutput::~MediaPlayer2AudioOutput() {
+ close();
+ free(mAttributes);
+ delete mCallbackData;
+}
+
+//static
+void MediaPlayer2AudioOutput::setMinBufferCount() {
+ char value[PROPERTY_VALUE_MAX];
+ if (property_get("ro.kernel.qemu", value, 0)) {
+ mIsOnEmulator = true;
+ mMinBufferCount = 12; // to prevent systematic buffer underrun for emulator
+ }
+}
+
+// static
+bool MediaPlayer2AudioOutput::isOnEmulator() {
+ setMinBufferCount(); // benign race wrt other threads
+ return mIsOnEmulator;
+}
+
+// static
+int MediaPlayer2AudioOutput::getMinBufferCount() {
+ setMinBufferCount(); // benign race wrt other threads
+ return mMinBufferCount;
+}
+
+ssize_t MediaPlayer2AudioOutput::bufferSize() const {
+ Mutex::Autolock lock(mLock);
+ if (mTrack == 0) {
+ return NO_INIT;
+ }
+ return mTrack->frameCount() * mFrameSize;
+}
+
+ssize_t MediaPlayer2AudioOutput::frameCount() const {
+ Mutex::Autolock lock(mLock);
+ if (mTrack == 0) {
+ return NO_INIT;
+ }
+ return mTrack->frameCount();
+}
+
+ssize_t MediaPlayer2AudioOutput::channelCount() const {
+ Mutex::Autolock lock(mLock);
+ if (mTrack == 0) {
+ return NO_INIT;
+ }
+ return mTrack->channelCount();
+}
+
+ssize_t MediaPlayer2AudioOutput::frameSize() const {
+ Mutex::Autolock lock(mLock);
+ if (mTrack == 0) {
+ return NO_INIT;
+ }
+ return mFrameSize;
+}
+
+uint32_t MediaPlayer2AudioOutput::latency () const {
+ Mutex::Autolock lock(mLock);
+ if (mTrack == 0) {
+ return 0;
+ }
+ return mTrack->latency();
+}
+
+float MediaPlayer2AudioOutput::msecsPerFrame() const {
+ Mutex::Autolock lock(mLock);
+ return mMsecsPerFrame;
+}
+
+status_t MediaPlayer2AudioOutput::getPosition(uint32_t *position) const {
+ Mutex::Autolock lock(mLock);
+ if (mTrack == 0) {
+ return NO_INIT;
+ }
+ return mTrack->getPosition(position);
+}
+
+status_t MediaPlayer2AudioOutput::getTimestamp(AudioTimestamp &ts) const {
+ Mutex::Autolock lock(mLock);
+ if (mTrack == 0) {
+ return NO_INIT;
+ }
+ return mTrack->getTimestamp(ts);
+}
+
+// TODO: Remove unnecessary calls to getPlayedOutDurationUs()
+// as it acquires locks and may query the audio driver.
+//
+// Some calls could conceivably retrieve extrapolated data instead of
+// accessing getTimestamp() or getPosition() every time a data buffer with
+// a media time is received.
+//
+// Calculate duration of played samples if played at normal rate (i.e., 1.0).
+int64_t MediaPlayer2AudioOutput::getPlayedOutDurationUs(int64_t nowUs) const {
+ Mutex::Autolock lock(mLock);
+ if (mTrack == 0 || mSampleRateHz == 0) {
+ return 0;
+ }
+
+ uint32_t numFramesPlayed;
+ int64_t numFramesPlayedAtUs;
+ AudioTimestamp ts;
+
+ status_t res = mTrack->getTimestamp(ts);
+ if (res == OK) { // case 1: mixing audio tracks and offloaded tracks.
+ numFramesPlayed = ts.mPosition;
+ numFramesPlayedAtUs = ts.mTime.tv_sec * 1000000LL + ts.mTime.tv_nsec / 1000;
+ //ALOGD("getTimestamp: OK %d %lld", numFramesPlayed, (long long)numFramesPlayedAtUs);
+ } else if (res == WOULD_BLOCK) { // case 2: transitory state on start of a new track
+ numFramesPlayed = 0;
+ numFramesPlayedAtUs = nowUs;
+ //ALOGD("getTimestamp: WOULD_BLOCK %d %lld",
+ // numFramesPlayed, (long long)numFramesPlayedAtUs);
+ } else { // case 3: transitory at new track or audio fast tracks.
+ res = mTrack->getPosition(&numFramesPlayed);
+ CHECK_EQ(res, (status_t)OK);
+ numFramesPlayedAtUs = nowUs;
+ numFramesPlayedAtUs += 1000LL * mTrack->latency() / 2; /* XXX */
+ //ALOGD("getPosition: %u %lld", numFramesPlayed, (long long)numFramesPlayedAtUs);
+ }
+
+ // CHECK_EQ(numFramesPlayed & (1 << 31), 0); // can't be negative until 12.4 hrs, test
+ // TODO: remove the (int32_t) casting below as it may overflow at 12.4 hours.
+ int64_t durationUs = (int64_t)((int32_t)numFramesPlayed * 1000000LL / mSampleRateHz)
+ + nowUs - numFramesPlayedAtUs;
+ if (durationUs < 0) {
+ // Occurs when numFramesPlayed position is very small and the following:
+ // (1) In case 1, the time nowUs is computed before getTimestamp() is called and
+ // numFramesPlayedAtUs is greater than nowUs by time more than numFramesPlayed.
+ // (2) In case 3, using getPosition and adding mAudioSink->latency() to
+ // numFramesPlayedAtUs, by a time amount greater than numFramesPlayed.
+ //
+ // Both of these are transitory conditions.
+ ALOGV("getPlayedOutDurationUs: negative duration %lld set to zero", (long long)durationUs);
+ durationUs = 0;
+ }
+ ALOGV("getPlayedOutDurationUs(%lld) nowUs(%lld) frames(%u) framesAt(%lld)",
+ (long long)durationUs, (long long)nowUs,
+ numFramesPlayed, (long long)numFramesPlayedAtUs);
+ return durationUs;
+}
+
+status_t MediaPlayer2AudioOutput::getFramesWritten(uint32_t *frameswritten) const {
+ Mutex::Autolock lock(mLock);
+ if (mTrack == 0) {
+ return NO_INIT;
+ }
+ ExtendedTimestamp ets;
+ status_t status = mTrack->getTimestamp(&ets);
+ if (status == OK || status == WOULD_BLOCK) {
+ *frameswritten = (uint32_t)ets.mPosition[ExtendedTimestamp::LOCATION_CLIENT];
+ }
+ return status;
+}
+
+status_t MediaPlayer2AudioOutput::setParameters(const String8& keyValuePairs) {
+ Mutex::Autolock lock(mLock);
+ if (mTrack == 0) {
+ return NO_INIT;
+ }
+ return mTrack->setParameters(keyValuePairs);
+}
+
+String8 MediaPlayer2AudioOutput::getParameters(const String8& keys) {
+ Mutex::Autolock lock(mLock);
+ if (mTrack == 0) {
+ return String8::empty();
+ }
+ return mTrack->getParameters(keys);
+}
+
+void MediaPlayer2AudioOutput::setAudioAttributes(const audio_attributes_t * attributes) {
+ Mutex::Autolock lock(mLock);
+ if (attributes == NULL) {
+ free(mAttributes);
+ mAttributes = NULL;
+ } else {
+ if (mAttributes == NULL) {
+ mAttributes = (audio_attributes_t *) calloc(1, sizeof(audio_attributes_t));
+ }
+ memcpy(mAttributes, attributes, sizeof(audio_attributes_t));
+ mStreamType = audio_attributes_to_stream_type(attributes);
+ }
+}
+
+void MediaPlayer2AudioOutput::setAudioStreamType(audio_stream_type_t streamType) {
+ Mutex::Autolock lock(mLock);
+ // do not allow direct stream type modification if attributes have been set
+ if (mAttributes == NULL) {
+ mStreamType = streamType;
+ }
+}
+
+void MediaPlayer2AudioOutput::close_l() {
+ mTrack.clear();
+}
+
+status_t MediaPlayer2AudioOutput::open(
+ uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
+ audio_format_t format, int bufferCount,
+ AudioCallback cb, void *cookie,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo,
+ bool doNotReconnect,
+ uint32_t suggestedFrameCount) {
+ ALOGV("open(%u, %d, 0x%x, 0x%x, %d, %d 0x%x)", sampleRate, channelCount, channelMask,
+ format, bufferCount, mSessionId, flags);
+
+ // offloading is only supported in callback mode for now.
+ // offloadInfo must be present if offload flag is set
+ if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) &&
+ ((cb == NULL) || (offloadInfo == NULL))) {
+ return BAD_VALUE;
+ }
+
+ // compute frame count for the AudioTrack internal buffer
+ size_t frameCount;
+ if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+ frameCount = 0; // AudioTrack will get frame count from AudioFlinger
+ } else {
+ // try to estimate the buffer processing fetch size from AudioFlinger.
+ // framesPerBuffer is approximate and generally correct, except when it's not :-).
+ uint32_t afSampleRate;
+ size_t afFrameCount;
+ if (AudioSystem::getOutputFrameCount(&afFrameCount, mStreamType) != NO_ERROR) {
+ return NO_INIT;
+ }
+ if (AudioSystem::getOutputSamplingRate(&afSampleRate, mStreamType) != NO_ERROR) {
+ return NO_INIT;
+ }
+ const size_t framesPerBuffer =
+ (unsigned long long)sampleRate * afFrameCount / afSampleRate;
+
+ if (bufferCount == 0) {
+ // use suggestedFrameCount
+ bufferCount = (suggestedFrameCount + framesPerBuffer - 1) / framesPerBuffer;
+ }
+ // Check argument bufferCount against the mininum buffer count
+ if (bufferCount != 0 && bufferCount < mMinBufferCount) {
+ ALOGV("bufferCount (%d) increased to %d", bufferCount, mMinBufferCount);
+ bufferCount = mMinBufferCount;
+ }
+ // if frameCount is 0, then AudioTrack will get frame count from AudioFlinger
+ // which will be the minimum size permitted.
+ frameCount = bufferCount * framesPerBuffer;
+ }
+
+ if (channelMask == CHANNEL_MASK_USE_CHANNEL_ORDER) {
+ channelMask = audio_channel_out_mask_from_count(channelCount);
+ if (0 == channelMask) {
+ ALOGE("open() error, can\'t derive mask for %d audio channels", channelCount);
+ return NO_INIT;
+ }
+ }
+
+ Mutex::Autolock lock(mLock);
+ mCallback = cb;
+ mCallbackCookie = cookie;
+
+ sp<AudioTrack> t;
+ CallbackData *newcbd = NULL;
+
+ ALOGV("creating new AudioTrack");
+
+ if (mCallback != NULL) {
+ newcbd = new CallbackData(this);
+ t = new AudioTrack(
+ mStreamType,
+ sampleRate,
+ format,
+ channelMask,
+ frameCount,
+ flags,
+ CallbackWrapper,
+ newcbd,
+ 0, // notification frames
+ mSessionId,
+ AudioTrack::TRANSFER_CALLBACK,
+ offloadInfo,
+ mUid,
+ mPid,
+ mAttributes,
+ doNotReconnect,
+ 1.0f, // default value for maxRequiredSpeed
+ mSelectedDeviceId);
+ } else {
+ // TODO: Due to buffer memory concerns, we use a max target playback speed
+ // based on mPlaybackRate at the time of open (instead of kMaxRequiredSpeed),
+ // also clamping the target speed to 1.0 <= targetSpeed <= kMaxRequiredSpeed.
+ const float targetSpeed =
+ std::min(std::max(mPlaybackRate.mSpeed, 1.0f), kMaxRequiredSpeed);
+ ALOGW_IF(targetSpeed != mPlaybackRate.mSpeed,
+ "track target speed:%f clamped from playback speed:%f",
+ targetSpeed, mPlaybackRate.mSpeed);
+ t = new AudioTrack(
+ mStreamType,
+ sampleRate,
+ format,
+ channelMask,
+ frameCount,
+ flags,
+ NULL, // callback
+ NULL, // user data
+ 0, // notification frames
+ mSessionId,
+ AudioTrack::TRANSFER_DEFAULT,
+ NULL, // offload info
+ mUid,
+ mPid,
+ mAttributes,
+ doNotReconnect,
+ targetSpeed,
+ mSelectedDeviceId);
+ }
+
+ if ((t == 0) || (t->initCheck() != NO_ERROR)) {
+ ALOGE("Unable to create audio track");
+ delete newcbd;
+ // t goes out of scope, so reference count drops to zero
+ return NO_INIT;
+ } else {
+ // successful AudioTrack initialization implies a legacy stream type was generated
+ // from the audio attributes
+ mStreamType = t->streamType();
+ }
+
+ CHECK((t != NULL) && ((mCallback == NULL) || (newcbd != NULL)));
+
+ mCallbackData = newcbd;
+ ALOGV("setVolume");
+ t->setVolume(mLeftVolume, mRightVolume);
+
+ mSampleRateHz = sampleRate;
+ mFlags = flags;
+ mMsecsPerFrame = 1E3f / (mPlaybackRate.mSpeed * sampleRate);
+ mFrameSize = t->frameSize();
+ mTrack = t;
+
+ return updateTrack_l();
+}
+
+status_t MediaPlayer2AudioOutput::updateTrack_l() {
+ if (mTrack == NULL) {
+ return NO_ERROR;
+ }
+
+ status_t res = NO_ERROR;
+ // Note some output devices may give us a direct track even though we don't specify it.
+ // Example: Line application b/17459982.
+ if ((mTrack->getFlags()
+ & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT)) == 0) {
+ res = mTrack->setPlaybackRate(mPlaybackRate);
+ if (res == NO_ERROR) {
+ mTrack->setAuxEffectSendLevel(mSendLevel);
+ res = mTrack->attachAuxEffect(mAuxEffectId);
+ }
+ }
+ mTrack->setOutputDevice(mSelectedDeviceId);
+ if (mDeviceCallbackEnabled) {
+ mTrack->addAudioDeviceCallback(mDeviceCallback.promote());
+ }
+ ALOGV("updateTrack_l() DONE status %d", res);
+ return res;
+}
+
+status_t MediaPlayer2AudioOutput::start() {
+ ALOGV("start");
+ Mutex::Autolock lock(mLock);
+ if (mCallbackData != NULL) {
+ mCallbackData->endTrackSwitch();
+ }
+ if (mTrack != 0) {
+ mTrack->setVolume(mLeftVolume, mRightVolume);
+ mTrack->setAuxEffectSendLevel(mSendLevel);
+ status_t status = mTrack->start();
+ return status;
+ }
+ return NO_INIT;
+}
+
+ssize_t MediaPlayer2AudioOutput::write(const void* buffer, size_t size, bool blocking) {
+ Mutex::Autolock lock(mLock);
+ LOG_ALWAYS_FATAL_IF(mCallback != NULL, "Don't call write if supplying a callback.");
+
+ //ALOGV("write(%p, %u)", buffer, size);
+ if (mTrack != 0) {
+ return mTrack->write(buffer, size, blocking);
+ }
+ return NO_INIT;
+}
+
+void MediaPlayer2AudioOutput::stop() {
+ ALOGV("stop");
+ Mutex::Autolock lock(mLock);
+ if (mTrack != 0) {
+ mTrack->stop();
+ }
+}
+
+void MediaPlayer2AudioOutput::flush() {
+ ALOGV("flush");
+ Mutex::Autolock lock(mLock);
+ if (mTrack != 0) {
+ mTrack->flush();
+ }
+}
+
+void MediaPlayer2AudioOutput::pause() {
+ ALOGV("pause");
+ Mutex::Autolock lock(mLock);
+ if (mTrack != 0) {
+ mTrack->pause();
+ }
+}
+
+void MediaPlayer2AudioOutput::close() {
+ ALOGV("close");
+ sp<AudioTrack> track;
+ {
+ Mutex::Autolock lock(mLock);
+ track = mTrack;
+ close_l(); // clears mTrack
+ }
+ // destruction of the track occurs outside of mutex.
+}
+
+void MediaPlayer2AudioOutput::setVolume(float left, float right) {
+ ALOGV("setVolume(%f, %f)", left, right);
+ Mutex::Autolock lock(mLock);
+ mLeftVolume = left;
+ mRightVolume = right;
+ if (mTrack != 0) {
+ mTrack->setVolume(left, right);
+ }
+}
+
+status_t MediaPlayer2AudioOutput::setPlaybackRate(const AudioPlaybackRate &rate) {
+ ALOGV("setPlaybackRate(%f %f %d %d)",
+ rate.mSpeed, rate.mPitch, rate.mFallbackMode, rate.mStretchMode);
+ Mutex::Autolock lock(mLock);
+ if (mTrack == 0) {
+ // remember rate so that we can set it when the track is opened
+ mPlaybackRate = rate;
+ return OK;
+ }
+ status_t res = mTrack->setPlaybackRate(rate);
+ if (res != NO_ERROR) {
+ return res;
+ }
+ // rate.mSpeed is always greater than 0 if setPlaybackRate succeeded
+ CHECK_GT(rate.mSpeed, 0.f);
+ mPlaybackRate = rate;
+ if (mSampleRateHz != 0) {
+ mMsecsPerFrame = 1E3f / (rate.mSpeed * mSampleRateHz);
+ }
+ return res;
+}
+
+status_t MediaPlayer2AudioOutput::getPlaybackRate(AudioPlaybackRate *rate) {
+ ALOGV("setPlaybackRate");
+ Mutex::Autolock lock(mLock);
+ if (mTrack == 0) {
+ return NO_INIT;
+ }
+ *rate = mTrack->getPlaybackRate();
+ return NO_ERROR;
+}
+
+status_t MediaPlayer2AudioOutput::setAuxEffectSendLevel(float level) {
+ ALOGV("setAuxEffectSendLevel(%f)", level);
+ Mutex::Autolock lock(mLock);
+ mSendLevel = level;
+ if (mTrack != 0) {
+ return mTrack->setAuxEffectSendLevel(level);
+ }
+ return NO_ERROR;
+}
+
+status_t MediaPlayer2AudioOutput::attachAuxEffect(int effectId) {
+ ALOGV("attachAuxEffect(%d)", effectId);
+ Mutex::Autolock lock(mLock);
+ mAuxEffectId = effectId;
+ if (mTrack != 0) {
+ return mTrack->attachAuxEffect(effectId);
+ }
+ return NO_ERROR;
+}
+
+status_t MediaPlayer2AudioOutput::setOutputDevice(audio_port_handle_t deviceId) {
+ ALOGV("setOutputDevice(%d)", deviceId);
+ Mutex::Autolock lock(mLock);
+ mSelectedDeviceId = deviceId;
+ if (mTrack != 0) {
+ return mTrack->setOutputDevice(deviceId);
+ }
+ return NO_ERROR;
+}
+
+status_t MediaPlayer2AudioOutput::getRoutedDeviceId(audio_port_handle_t* deviceId) {
+ ALOGV("getRoutedDeviceId");
+ Mutex::Autolock lock(mLock);
+ if (mTrack != 0) {
+ mRoutedDeviceId = mTrack->getRoutedDeviceId();
+ }
+ *deviceId = mRoutedDeviceId;
+ return NO_ERROR;
+}
+
+status_t MediaPlayer2AudioOutput::enableAudioDeviceCallback(bool enabled) {
+ ALOGV("enableAudioDeviceCallback, %d", enabled);
+ Mutex::Autolock lock(mLock);
+ mDeviceCallbackEnabled = enabled;
+ if (mTrack != 0) {
+ status_t status;
+ if (enabled) {
+ status = mTrack->addAudioDeviceCallback(mDeviceCallback.promote());
+ } else {
+ status = mTrack->removeAudioDeviceCallback(mDeviceCallback.promote());
+ }
+ return status;
+ }
+ return NO_ERROR;
+}
+
+// static
+void MediaPlayer2AudioOutput::CallbackWrapper(
+ int event, void *cookie, void *info) {
+ //ALOGV("callbackwrapper");
+ CallbackData *data = (CallbackData*)cookie;
+ // lock to ensure we aren't caught in the middle of a track switch.
+ data->lock();
+ MediaPlayer2AudioOutput *me = data->getOutput();
+ AudioTrack::Buffer *buffer = (AudioTrack::Buffer *)info;
+ if (me == NULL) {
+ // no output set, likely because the track was scheduled to be reused
+ // by another player, but the format turned out to be incompatible.
+ data->unlock();
+ if (buffer != NULL) {
+ buffer->size = 0;
+ }
+ return;
+ }
+
+ switch(event) {
+ case AudioTrack::EVENT_MORE_DATA: {
+ size_t actualSize = (*me->mCallback)(
+ me, buffer->raw, buffer->size, me->mCallbackCookie,
+ CB_EVENT_FILL_BUFFER);
+
+ // Log when no data is returned from the callback.
+ // (1) We may have no data (especially with network streaming sources).
+ // (2) We may have reached the EOS and the audio track is not stopped yet.
+ // Note that AwesomePlayer/AudioPlayer will only return zero size when it reaches the EOS.
+ // NuPlayer2Renderer will return zero when it doesn't have data (it doesn't block to fill).
+ //
+ // This is a benign busy-wait, with the next data request generated 10 ms or more later;
+ // nevertheless for power reasons, we don't want to see too many of these.
+
+ ALOGV_IF(actualSize == 0 && buffer->size > 0, "callbackwrapper: empty buffer returned");
+
+ buffer->size = actualSize;
+ } break;
+
+ case AudioTrack::EVENT_STREAM_END:
+ // currently only occurs for offloaded callbacks
+ ALOGV("callbackwrapper: deliver EVENT_STREAM_END");
+ (*me->mCallback)(me, NULL /* buffer */, 0 /* size */,
+ me->mCallbackCookie, CB_EVENT_STREAM_END);
+ break;
+
+ case AudioTrack::EVENT_NEW_IAUDIOTRACK :
+ ALOGV("callbackwrapper: deliver EVENT_TEAR_DOWN");
+ (*me->mCallback)(me, NULL /* buffer */, 0 /* size */,
+ me->mCallbackCookie, CB_EVENT_TEAR_DOWN);
+ break;
+
+ case AudioTrack::EVENT_UNDERRUN:
+ // This occurs when there is no data available, typically
+ // when there is a failure to supply data to the AudioTrack. It can also
+ // occur in non-offloaded mode when the audio device comes out of standby.
+ //
+ // If an AudioTrack underruns it outputs silence. Since this happens suddenly
+ // it may sound like an audible pop or glitch.
+ //
+ // The underrun event is sent once per track underrun; the condition is reset
+ // when more data is sent to the AudioTrack.
+ ALOGD("callbackwrapper: EVENT_UNDERRUN (discarded)");
+ break;
+
+ default:
+ ALOGE("received unknown event type: %d inside CallbackWrapper !", event);
+ }
+
+ data->unlock();
+}
+
+audio_session_t MediaPlayer2AudioOutput::getSessionId() const
+{
+ Mutex::Autolock lock(mLock);
+ return mSessionId;
+}
+
+uint32_t MediaPlayer2AudioOutput::getSampleRate() const
+{
+ Mutex::Autolock lock(mLock);
+ if (mTrack == 0) {
+ return 0;
+ }
+ return mTrack->getSampleRate();
+}
+
+int64_t MediaPlayer2AudioOutput::getBufferDurationInUs() const
+{
+ Mutex::Autolock lock(mLock);
+ if (mTrack == 0) {
+ return 0;
+ }
+ int64_t duration;
+ if (mTrack->getBufferDurationInUs(&duration) != OK) {
+ return 0;
+ }
+ return duration;
+}
+
+} // namespace android