Move AudioPlayer to test commands

AudioPlayer was only used by the commandline utilities, so move it
out of libstagefright.

Test: build, run
Change-Id: I561cccd323206de7415bd235b72711194080aaea

7 files changed, 828 insertions, 11 deletions

diff --git a/cmds/stagefright/Android.mk b/cmds/stagefright/Android.mk
index cc43b61612..defc94f8fd 100644
--- a/cmds/stagefright/Android.mk
+++ b/cmds/stagefright/Android.mk
@@ -3,6 +3,7 @@ LOCAL_PATH:= $(call my-dir)
 include $(CLEAR_VARS)
 
 LOCAL_SRC_FILES:=       \
+        AudioPlayer.cpp \
         stagefright.cpp \
         jpeg.cpp        \
         SineSource.cpp
@@ -10,7 +11,7 @@ LOCAL_SRC_FILES:=       \
 LOCAL_SHARED_LIBRARIES := \
         libstagefright libmedia libmedia_codeclist libutils libbinder \
         libstagefright_foundation libjpeg libui libgui libcutils liblog \
-        libhidlbase libdatasource \
+        libhidlbase libdatasource libaudioclient \
         android.hardware.media.omx@1.0 \
 
 LOCAL_C_INCLUDES:= \
@@ -31,12 +32,13 @@ include $(BUILD_EXECUTABLE)
 include $(CLEAR_VARS)
 
 LOCAL_SRC_FILES:=         \
+        AudioPlayer.cpp \
         SineSource.cpp    \
         record.cpp
 
 LOCAL_SHARED_LIBRARIES := \
         libstagefright libmedia liblog libutils libbinder \
-        libstagefright_foundation libdatasource
+        libstagefright_foundation libdatasource libaudioclient
 
 LOCAL_C_INCLUDES:= \
         frameworks/av/camera/include \
@@ -57,12 +59,12 @@ include $(BUILD_EXECUTABLE)
 include $(CLEAR_VARS)
 
 LOCAL_SRC_FILES:=         \
-        SineSource.cpp    \
+        AudioPlayer.cpp \
         recordvideo.cpp
 
 LOCAL_SHARED_LIBRARIES := \
         libstagefright libmedia liblog libutils libbinder \
-        libstagefright_foundation
+        libstagefright_foundation libaudioclient
 
 LOCAL_C_INCLUDES:= \
         frameworks/av/media/libstagefright \
@@ -83,12 +85,13 @@ include $(BUILD_EXECUTABLE)
 include $(CLEAR_VARS)
 
 LOCAL_SRC_FILES:=         \
+        AudioPlayer.cpp \
         SineSource.cpp    \
         audioloop.cpp
 
 LOCAL_SHARED_LIBRARIES := \
         libstagefright libmedia liblog libutils libbinder \
-        libstagefright_foundation
+        libstagefright_foundation libaudioclient
 
 LOCAL_C_INCLUDES:= \
         frameworks/av/media/libstagefright \
diff --git a/cmds/stagefright/AudioPlayer.cpp b/cmds/stagefright/AudioPlayer.cpp
new file mode 100644
index 0000000000..208713d950
--- /dev/null
+++ b/cmds/stagefright/AudioPlayer.cpp
@@ -0,0 +1,689 @@
+/*
+ * Copyright (C) 2009 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.
+ */
+
+#include <inttypes.h>
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "AudioPlayer"
+#include <utils/Log.h>
+#include <cutils/compiler.h>
+
+#include <binder/IPCThreadState.h>
+#include <media/AudioTrack.h>
+#include <media/MediaSource.h>
+#include <media/openmax/OMX_Audio.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/ALookup.h>
+#include <media/stagefright/foundation/ALooper.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/Utils.h>
+
+#include "AudioPlayer.h"
+
+namespace android {
+
+AudioPlayer::AudioPlayer(
+        const sp<MediaPlayerBase::AudioSink> &audioSink,
+        uint32_t flags)
+    : mInputBuffer(NULL),
+      mSampleRate(0),
+      mLatencyUs(0),
+      mFrameSize(0),
+      mNumFramesPlayed(0),
+      mNumFramesPlayedSysTimeUs(ALooper::GetNowUs()),
+      mPositionTimeMediaUs(-1),
+      mPositionTimeRealUs(-1),
+      mSeeking(false),
+      mReachedEOS(false),
+      mFinalStatus(OK),
+      mSeekTimeUs(0),
+      mStarted(false),
+      mIsFirstBuffer(false),
+      mFirstBufferResult(OK),
+      mFirstBuffer(NULL),
+      mAudioSink(audioSink),
+      mPlaying(false),
+      mStartPosUs(0),
+      mCreateFlags(flags) {
+}
+
+AudioPlayer::~AudioPlayer() {
+    if (mStarted) {
+        reset();
+    }
+}
+
+void AudioPlayer::setSource(const sp<MediaSource> &source) {
+    CHECK(mSource == NULL);
+    mSource = source;
+}
+
+ALookup<audio_format_t, int32_t> sAudioFormatToPcmEncoding {
+    {
+        { AUDIO_FORMAT_PCM_16_BIT, kAudioEncodingPcm16bit },
+        { AUDIO_FORMAT_PCM_8_BIT,  kAudioEncodingPcm8bit  },
+        { AUDIO_FORMAT_PCM_FLOAT,  kAudioEncodingPcmFloat },
+    }
+};
+
+status_t AudioPlayer::start(bool sourceAlreadyStarted) {
+    CHECK(!mStarted);
+    CHECK(mSource != NULL);
+
+    status_t err;
+    if (!sourceAlreadyStarted) {
+        err = mSource->start();
+
+        if (err != OK) {
+            return err;
+        }
+    }
+
+    // We allow an optional INFO_FORMAT_CHANGED at the very beginning
+    // of playback, if there is one, getFormat below will retrieve the
+    // updated format, if there isn't, we'll stash away the valid buffer
+    // of data to be used on the first audio callback.
+
+    CHECK(mFirstBuffer == NULL);
+
+    MediaSource::ReadOptions options;
+    if (mSeeking) {
+        options.setSeekTo(mSeekTimeUs);
+        mSeeking = false;
+    }
+
+    mFirstBufferResult = mSource->read(&mFirstBuffer, &options);
+    if (mFirstBufferResult == INFO_FORMAT_CHANGED) {
+        ALOGV("INFO_FORMAT_CHANGED!!!");
+
+        CHECK(mFirstBuffer == NULL);
+        mFirstBufferResult = OK;
+        mIsFirstBuffer = false;
+    } else {
+        mIsFirstBuffer = true;
+    }
+
+    sp<MetaData> format = mSource->getFormat();
+
+    if (format == NULL) {
+        ALOGE("No metadata b/118620871");
+        android_errorWriteLog(0x534e4554, "118620871");
+        return BAD_VALUE;
+    }
+
+    const char *mime;
+    bool success = format->findCString(kKeyMIMEType, &mime);
+    CHECK(success);
+    CHECK(useOffload() || !strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_RAW));
+
+    success = format->findInt32(kKeySampleRate, &mSampleRate);
+    CHECK(success);
+
+    int32_t numChannels, channelMask;
+    success = format->findInt32(kKeyChannelCount, &numChannels);
+    CHECK(success);
+
+    if(!format->findInt32(kKeyChannelMask, &channelMask)) {
+        // log only when there's a risk of ambiguity of channel mask selection
+        ALOGI_IF(numChannels > 2,
+                "source format didn't specify channel mask, using (%d) channel order", numChannels);
+        channelMask = CHANNEL_MASK_USE_CHANNEL_ORDER;
+    }
+
+    audio_format_t audioFormat = AUDIO_FORMAT_PCM_16_BIT;
+    int32_t pcmEncoding;
+    if (format->findInt32(kKeyPcmEncoding, &pcmEncoding)) {
+        sAudioFormatToPcmEncoding.map(pcmEncoding, &audioFormat);
+    }
+
+    if (useOffload()) {
+        if (mapMimeToAudioFormat(audioFormat, mime) != OK) {
+            ALOGE("Couldn't map mime type \"%s\" to a valid AudioSystem::audio_format", mime);
+            audioFormat = AUDIO_FORMAT_INVALID;
+        } else {
+            ALOGV("Mime type \"%s\" mapped to audio_format 0x%x", mime, audioFormat);
+        }
+
+        int32_t aacaot = -1;
+        if ((audioFormat == AUDIO_FORMAT_AAC) && format->findInt32(kKeyAACAOT, &aacaot)) {
+            // Redefine AAC format corrosponding to aac profile
+            mapAACProfileToAudioFormat(audioFormat,(OMX_AUDIO_AACPROFILETYPE) aacaot);
+        }
+    }
+
+    int avgBitRate = -1;
+    format->findInt32(kKeyBitRate, &avgBitRate);
+
+    if (mAudioSink.get() != NULL) {
+
+        uint32_t flags = AUDIO_OUTPUT_FLAG_NONE;
+        audio_offload_info_t offloadInfo = AUDIO_INFO_INITIALIZER;
+
+        if (allowDeepBuffering()) {
+            flags |= AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
+        }
+        if (useOffload()) {
+            flags |= AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD;
+
+            int64_t durationUs;
+            if (format->findInt64(kKeyDuration, &durationUs)) {
+                offloadInfo.duration_us = durationUs;
+            } else {
+                offloadInfo.duration_us = -1;
+            }
+
+            offloadInfo.sample_rate = mSampleRate;
+            offloadInfo.channel_mask = channelMask;
+            offloadInfo.format = audioFormat;
+            offloadInfo.stream_type = AUDIO_STREAM_MUSIC;
+            offloadInfo.bit_rate = avgBitRate;
+            offloadInfo.has_video = ((mCreateFlags & HAS_VIDEO) != 0);
+            offloadInfo.is_streaming = ((mCreateFlags & IS_STREAMING) != 0);
+        }
+
+        status_t err = mAudioSink->open(
+                mSampleRate, numChannels, channelMask, audioFormat,
+                DEFAULT_AUDIOSINK_BUFFERCOUNT,
+                &AudioPlayer::AudioSinkCallback,
+                this,
+                (audio_output_flags_t)flags,
+                useOffload() ? &offloadInfo : NULL);
+
+        if (err == OK) {
+            mLatencyUs = (int64_t)mAudioSink->latency() * 1000;
+            mFrameSize = mAudioSink->frameSize();
+
+            if (useOffload()) {
+                // If the playback is offloaded to h/w we pass the
+                // HAL some metadata information
+                // We don't want to do this for PCM because it will be going
+                // through the AudioFlinger mixer before reaching the hardware
+                sendMetaDataToHal(mAudioSink, format);
+            }
+
+            err = mAudioSink->start();
+            // do not alter behavior for non offloaded tracks: ignore start status.
+            if (!useOffload()) {
+                err = OK;
+            }
+        }
+
+        if (err != OK) {
+            if (mFirstBuffer != NULL) {
+                mFirstBuffer->release();
+                mFirstBuffer = NULL;
+            }
+
+            if (!sourceAlreadyStarted) {
+                mSource->stop();
+            }
+
+            return err;
+        }
+
+    } else {
+        // playing to an AudioTrack, set up mask if necessary
+        audio_channel_mask_t audioMask = channelMask == CHANNEL_MASK_USE_CHANNEL_ORDER ?
+                audio_channel_out_mask_from_count(numChannels) : channelMask;
+        if (0 == audioMask) {
+            return BAD_VALUE;
+        }
+
+        mAudioTrack = new AudioTrack(
+                AUDIO_STREAM_MUSIC, mSampleRate, AUDIO_FORMAT_PCM_16_BIT, audioMask,
+                0 /*frameCount*/, AUDIO_OUTPUT_FLAG_NONE, &AudioCallback, this,
+                0 /*notificationFrames*/);
+
+        if ((err = mAudioTrack->initCheck()) != OK) {
+            mAudioTrack.clear();
+
+            if (mFirstBuffer != NULL) {
+                mFirstBuffer->release();
+                mFirstBuffer = NULL;
+            }
+
+            if (!sourceAlreadyStarted) {
+                mSource->stop();
+            }
+
+            return err;
+        }
+
+        mLatencyUs = (int64_t)mAudioTrack->latency() * 1000;
+        mFrameSize = mAudioTrack->frameSize();
+
+        mAudioTrack->start();
+    }
+
+    mStarted = true;
+    mPlaying = true;
+
+    return OK;
+}
+
+void AudioPlayer::pause(bool playPendingSamples) {
+    CHECK(mStarted);
+
+    if (playPendingSamples) {
+        if (mAudioSink.get() != NULL) {
+            mAudioSink->stop();
+        } else {
+            mAudioTrack->stop();
+        }
+
+        mNumFramesPlayed = 0;
+        mNumFramesPlayedSysTimeUs = ALooper::GetNowUs();
+    } else {
+        if (mAudioSink.get() != NULL) {
+            mAudioSink->pause();
+        } else {
+            mAudioTrack->pause();
+        }
+    }
+
+    mPlaying = false;
+}
+
+status_t AudioPlayer::resume() {
+    CHECK(mStarted);
+    status_t err;
+
+    if (mAudioSink.get() != NULL) {
+        err = mAudioSink->start();
+    } else {
+        err = mAudioTrack->start();
+    }
+
+    if (err == OK) {
+        mPlaying = true;
+    }
+
+    return err;
+}
+
+void AudioPlayer::reset() {
+    CHECK(mStarted);
+
+    ALOGV("reset: mPlaying=%d mReachedEOS=%d useOffload=%d",
+                                mPlaying, mReachedEOS, useOffload() );
+
+    if (mAudioSink.get() != NULL) {
+        mAudioSink->stop();
+        // If we're closing and have reached EOS, we don't want to flush
+        // the track because if it is offloaded there could be a small
+        // amount of residual data in the hardware buffer which we must
+        // play to give gapless playback.
+        // But if we're resetting when paused or before we've reached EOS
+        // we can't be doing a gapless playback and there could be a large
+        // amount of data queued in the hardware if the track is offloaded,
+        // so we must flush to prevent a track switch being delayed playing
+        // the buffered data that we don't want now
+        if (!mPlaying || !mReachedEOS) {
+            mAudioSink->flush();
+        }
+
+        mAudioSink->close();
+    } else {
+        mAudioTrack->stop();
+
+        if (!mPlaying || !mReachedEOS) {
+            mAudioTrack->flush();
+        }
+
+        mAudioTrack.clear();
+    }
+
+    // Make sure to release any buffer we hold onto so that the
+    // source is able to stop().
+
+    if (mFirstBuffer != NULL) {
+        mFirstBuffer->release();
+        mFirstBuffer = NULL;
+    }
+
+    if (mInputBuffer != NULL) {
+        ALOGV("AudioPlayer releasing input buffer.");
+
+        mInputBuffer->release();
+        mInputBuffer = NULL;
+    }
+
+    mSource->stop();
+
+    // The following hack is necessary to ensure that the OMX
+    // component is completely released by the time we may try
+    // to instantiate it again.
+    // When offloading, the OMX component is not used so this hack
+    // is not needed
+    if (!useOffload()) {
+        wp<MediaSource> tmp = mSource;
+        mSource.clear();
+        while (tmp.promote() != NULL) {
+            usleep(1000);
+        }
+    } else {
+        mSource.clear();
+    }
+    IPCThreadState::self()->flushCommands();
+
+    mNumFramesPlayed = 0;
+    mNumFramesPlayedSysTimeUs = ALooper::GetNowUs();
+    mPositionTimeMediaUs = -1;
+    mPositionTimeRealUs = -1;
+    mSeeking = false;
+    mSeekTimeUs = 0;
+    mReachedEOS = false;
+    mFinalStatus = OK;
+    mStarted = false;
+    mPlaying = false;
+    mStartPosUs = 0;
+}
+
+// static
+void AudioPlayer::AudioCallback(int event, void *user, void *info) {
+    static_cast<AudioPlayer *>(user)->AudioCallback(event, info);
+}
+
+bool AudioPlayer::reachedEOS(status_t *finalStatus) {
+    *finalStatus = OK;
+
+    Mutex::Autolock autoLock(mLock);
+    *finalStatus = mFinalStatus;
+    return mReachedEOS;
+}
+
+status_t AudioPlayer::setPlaybackRate(const AudioPlaybackRate &rate) {
+    if (mAudioSink.get() != NULL) {
+        return mAudioSink->setPlaybackRate(rate);
+    } else if (mAudioTrack != 0){
+        return mAudioTrack->setPlaybackRate(rate);
+    } else {
+        return NO_INIT;
+    }
+}
+
+status_t AudioPlayer::getPlaybackRate(AudioPlaybackRate *rate /* nonnull */) {
+    if (mAudioSink.get() != NULL) {
+        return mAudioSink->getPlaybackRate(rate);
+    } else if (mAudioTrack != 0) {
+        *rate = mAudioTrack->getPlaybackRate();
+        return OK;
+    } else {
+        return NO_INIT;
+    }
+}
+
+// static
+size_t AudioPlayer::AudioSinkCallback(
+        MediaPlayerBase::AudioSink * /* audioSink */,
+        void *buffer, size_t size, void *cookie,
+        MediaPlayerBase::AudioSink::cb_event_t event) {
+    AudioPlayer *me = (AudioPlayer *)cookie;
+
+    switch(event) {
+    case MediaPlayerBase::AudioSink::CB_EVENT_FILL_BUFFER:
+        return me->fillBuffer(buffer, size);
+
+    case MediaPlayerBase::AudioSink::CB_EVENT_STREAM_END:
+        ALOGV("AudioSinkCallback: stream end");
+        me->mReachedEOS = true;
+        break;
+
+    case MediaPlayerBase::AudioSink::CB_EVENT_TEAR_DOWN:
+        ALOGV("AudioSinkCallback: Tear down event");
+        break;
+    }
+
+    return 0;
+}
+
+void AudioPlayer::AudioCallback(int event, void *info) {
+    switch (event) {
+    case AudioTrack::EVENT_MORE_DATA:
+        {
+        AudioTrack::Buffer *buffer = (AudioTrack::Buffer *)info;
+        size_t numBytesWritten = fillBuffer(buffer->raw, buffer->size);
+        buffer->size = numBytesWritten;
+        }
+        break;
+
+    case AudioTrack::EVENT_STREAM_END:
+        mReachedEOS = true;
+        break;
+    }
+}
+
+size_t AudioPlayer::fillBuffer(void *data, size_t size) {
+    if (mNumFramesPlayed == 0) {
+        ALOGV("AudioCallback");
+    }
+
+    if (mReachedEOS) {
+        return 0;
+    }
+
+    size_t size_done = 0;
+    size_t size_remaining = size;
+    while (size_remaining > 0) {
+        MediaSource::ReadOptions options;
+        bool refreshSeekTime = false;
+
+        {
+            Mutex::Autolock autoLock(mLock);
+
+            if (mSeeking) {
+                if (mIsFirstBuffer) {
+                    if (mFirstBuffer != NULL) {
+                        mFirstBuffer->release();
+                        mFirstBuffer = NULL;
+                    }
+                    mIsFirstBuffer = false;
+                }
+
+                options.setSeekTo(mSeekTimeUs);
+                refreshSeekTime = true;
+
+                if (mInputBuffer != NULL) {
+                    mInputBuffer->release();
+                    mInputBuffer = NULL;
+                }
+
+                mSeeking = false;
+            }
+        }
+
+        if (mInputBuffer == NULL) {
+            status_t err;
+
+            if (mIsFirstBuffer) {
+                mInputBuffer = mFirstBuffer;
+                mFirstBuffer = NULL;
+                err = mFirstBufferResult;
+
+                mIsFirstBuffer = false;
+            } else {
+                err = mSource->read(&mInputBuffer, &options);
+            }
+
+            CHECK((err == OK && mInputBuffer != NULL)
+                   || (err != OK && mInputBuffer == NULL));
+
+            Mutex::Autolock autoLock(mLock);
+
+            if (err != OK) {
+                if (!mReachedEOS) {
+                    if (useOffload()) {
+                        // no more buffers to push - stop() and wait for STREAM_END
+                        // don't set mReachedEOS until stream end received
+                        if (mAudioSink != NULL) {
+                            mAudioSink->stop();
+                        } else {
+                            mAudioTrack->stop();
+                        }
+                    } else {
+                        mReachedEOS = true;
+                    }
+                }
+
+                mFinalStatus = err;
+                break;
+            }
+
+            if (mAudioSink != NULL) {
+                mLatencyUs = (int64_t)mAudioSink->latency() * 1000;
+            } else {
+                mLatencyUs = (int64_t)mAudioTrack->latency() * 1000;
+            }
+
+            if(mInputBuffer->range_length() != 0) {
+                CHECK(mInputBuffer->meta_data().findInt64(
+                        kKeyTime, &mPositionTimeMediaUs));
+            }
+
+            // need to adjust the mStartPosUs for offload decoding since parser
+            // might not be able to get the exact seek time requested.
+            if (refreshSeekTime) {
+                if (useOffload()) {
+                    mStartPosUs = mPositionTimeMediaUs;
+                    ALOGV("adjust seek time to: %.2f", mStartPosUs/ 1E6);
+                }
+                // clear seek time with mLock locked and once we have valid mPositionTimeMediaUs
+                // and mPositionTimeRealUs
+                // before clearing mSeekTimeUs check if a new seek request has been received while
+                // we were reading from the source with mLock released.
+                if (!mSeeking) {
+                    mSeekTimeUs = 0;
+                }
+            }
+
+            if (!useOffload()) {
+                mPositionTimeRealUs =
+                    ((mNumFramesPlayed + size_done / mFrameSize) * 1000000)
+                        / mSampleRate;
+                ALOGV("buffer->size() = %zu, "
+                     "mPositionTimeMediaUs=%.2f mPositionTimeRealUs=%.2f",
+                     mInputBuffer->range_length(),
+                     mPositionTimeMediaUs / 1E6, mPositionTimeRealUs / 1E6);
+            }
+
+        }
+
+        if (mInputBuffer->range_length() == 0) {
+            mInputBuffer->release();
+            mInputBuffer = NULL;
+
+            continue;
+        }
+
+        size_t copy = size_remaining;
+        if (copy > mInputBuffer->range_length()) {
+            copy = mInputBuffer->range_length();
+        }
+
+        memcpy((char *)data + size_done,
+               (const char *)mInputBuffer->data() + mInputBuffer->range_offset(),
+               copy);
+
+        mInputBuffer->set_range(mInputBuffer->range_offset() + copy,
+                                mInputBuffer->range_length() - copy);
+
+        size_done += copy;
+        size_remaining -= copy;
+    }
+
+    if (useOffload()) {
+        // We must ask the hardware what it has played
+        mPositionTimeRealUs = getOutputPlayPositionUs_l();
+        ALOGV("mPositionTimeMediaUs=%.2f mPositionTimeRealUs=%.2f",
+             mPositionTimeMediaUs / 1E6, mPositionTimeRealUs / 1E6);
+    }
+
+    {
+        Mutex::Autolock autoLock(mLock);
+        mNumFramesPlayed += size_done / mFrameSize;
+        mNumFramesPlayedSysTimeUs = ALooper::GetNowUs();
+    }
+
+    return size_done;
+}
+
+int64_t AudioPlayer::getOutputPlayPositionUs_l()
+{
+    uint32_t playedSamples = 0;
+    uint32_t sampleRate;
+    if (mAudioSink != NULL) {
+        mAudioSink->getPosition(&playedSamples);
+        sampleRate = mAudioSink->getSampleRate();
+    } else {
+        mAudioTrack->getPosition(&playedSamples);
+        sampleRate = mAudioTrack->getSampleRate();
+    }
+    if (sampleRate != 0) {
+        mSampleRate = sampleRate;
+    }
+
+    int64_t playedUs;
+    if (mSampleRate != 0) {
+        playedUs = (static_cast<int64_t>(playedSamples) * 1000000 ) / mSampleRate;
+    } else {
+        playedUs = 0;
+    }
+
+    // HAL position is relative to the first buffer we sent at mStartPosUs
+    const int64_t renderedDuration = mStartPosUs + playedUs;
+    ALOGV("getOutputPlayPositionUs_l %" PRId64, renderedDuration);
+    return renderedDuration;
+}
+
+status_t AudioPlayer::seekTo(int64_t time_us) {
+    Mutex::Autolock autoLock(mLock);
+
+    ALOGV("seekTo( %" PRId64 " )", time_us);
+
+    mSeeking = true;
+    mPositionTimeRealUs = mPositionTimeMediaUs = -1;
+    mReachedEOS = false;
+    mSeekTimeUs = time_us;
+    mStartPosUs = time_us;
+
+    // Flush resets the number of played frames
+    mNumFramesPlayed = 0;
+    mNumFramesPlayedSysTimeUs = ALooper::GetNowUs();
+
+    if (mAudioSink != NULL) {
+        if (mPlaying) {
+            mAudioSink->pause();
+        }
+        mAudioSink->flush();
+        if (mPlaying) {
+            mAudioSink->start();
+        }
+    } else {
+        if (mPlaying) {
+            mAudioTrack->pause();
+        }
+        mAudioTrack->flush();
+        if (mPlaying) {
+            mAudioTrack->start();
+        }
+    }
+
+    return OK;
+}
+
+}
diff --git a/cmds/stagefright/AudioPlayer.h b/cmds/stagefright/AudioPlayer.h
new file mode 100644
index 0000000000..7c2c36fc0e
--- /dev/null
+++ b/cmds/stagefright/AudioPlayer.h
@@ -0,0 +1,125 @@
+/*
+ * Copyright (C) 2009 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.
+ */
+
+#ifndef AUDIO_PLAYER_H_
+
+#define AUDIO_PLAYER_H_
+
+#include <media/MediaSource.h>
+#include <media/MediaPlayerInterface.h>
+#include <media/stagefright/MediaBuffer.h>
+#include <utils/threads.h>
+
+namespace android {
+
+struct AudioPlaybackRate;
+class AudioTrack;
+struct AwesomePlayer;
+
+class AudioPlayer {
+public:
+    enum {
+        REACHED_EOS,
+        SEEK_COMPLETE
+    };
+
+    enum {
+        ALLOW_DEEP_BUFFERING = 0x01,
+        USE_OFFLOAD = 0x02,
+        HAS_VIDEO   = 0x1000,
+        IS_STREAMING = 0x2000
+
+    };
+
+    AudioPlayer(const sp<MediaPlayerBase::AudioSink> &audioSink,
+                uint32_t flags = 0);
+
+    virtual ~AudioPlayer();
+
+    // Caller retains ownership of "source".
+    void setSource(const sp<MediaSource> &source);
+
+    status_t start(bool sourceAlreadyStarted = false);
+
+    void pause(bool playPendingSamples = false);
+    status_t resume();
+
+    status_t seekTo(int64_t time_us);
+
+    bool isSeeking();
+    bool reachedEOS(status_t *finalStatus);
+
+    status_t setPlaybackRate(const AudioPlaybackRate &rate);
+    status_t getPlaybackRate(AudioPlaybackRate *rate /* nonnull */);
+
+private:
+    sp<MediaSource> mSource;
+    sp<AudioTrack> mAudioTrack;
+
+    MediaBufferBase *mInputBuffer;
+
+    int mSampleRate;
+    int64_t mLatencyUs;
+    size_t mFrameSize;
+
+    Mutex mLock;
+    int64_t mNumFramesPlayed;
+    int64_t mNumFramesPlayedSysTimeUs;
+
+    int64_t mPositionTimeMediaUs;
+    int64_t mPositionTimeRealUs;
+
+    bool mSeeking;
+    bool mReachedEOS;
+    status_t mFinalStatus;
+    int64_t mSeekTimeUs;
+
+    bool mStarted;
+
+    bool mIsFirstBuffer;
+    status_t mFirstBufferResult;
+    MediaBufferBase *mFirstBuffer;
+
+    sp<MediaPlayerBase::AudioSink> mAudioSink;
+
+    bool mPlaying;
+    int64_t mStartPosUs;
+    const uint32_t mCreateFlags;
+
+    static void AudioCallback(int event, void *user, void *info);
+    void AudioCallback(int event, void *info);
+
+    static size_t AudioSinkCallback(
+            MediaPlayerBase::AudioSink *audioSink,
+            void *data, size_t size, void *me,
+            MediaPlayerBase::AudioSink::cb_event_t event);
+
+    size_t fillBuffer(void *data, size_t size);
+
+    void reset();
+
+    int64_t getOutputPlayPositionUs_l();
+
+    bool allowDeepBuffering() const { return (mCreateFlags & ALLOW_DEEP_BUFFERING) != 0; }
+    bool useOffload() const { return (mCreateFlags & USE_OFFLOAD) != 0; }
+
+    AudioPlayer(const AudioPlayer &);
+    AudioPlayer &operator=(const AudioPlayer &);
+};
+
+}  // namespace android
+
+#endif  // AUDIO_PLAYER_H_
diff --git a/cmds/stagefright/audioloop.cpp b/cmds/stagefright/audioloop.cpp
index d4f2e8d8d4..bd274d8f67 100644
--- a/cmds/stagefright/audioloop.cpp
+++ b/cmds/stagefright/audioloop.cpp
@@ -29,11 +29,11 @@
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/AMessage.h>
 #include <media/stagefright/AMRWriter.h>
-#include <media/stagefright/AudioPlayer.h>
 #include <media/stagefright/AudioSource.h>
 #include <media/stagefright/MediaCodecSource.h>
 #include <media/stagefright/MediaDefs.h>
 #include <media/stagefright/SimpleDecodingSource.h>
+#include "AudioPlayer.h"
 #include "SineSource.h"
 
 using namespace android;
diff --git a/cmds/stagefright/record.cpp b/cmds/stagefright/record.cpp
index 498237de8d..37091c43e1 100644
--- a/cmds/stagefright/record.cpp
+++ b/cmds/stagefright/record.cpp
@@ -21,7 +21,6 @@
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/ALooper.h>
 #include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/AudioPlayer.h>
 #include <media/stagefright/CameraSource.h>
 #include <media/stagefright/MediaBufferGroup.h>
 #include <media/stagefright/MediaDefs.h>
@@ -33,6 +32,8 @@
 #include <media/stagefright/SimpleDecodingSource.h>
 #include <media/MediaPlayerInterface.h>
 
+#include "AudioPlayer.h"
+
 using namespace android;
 
 static const int32_t kAudioBitRate = 12200;
diff --git a/cmds/stagefright/recordvideo.cpp b/cmds/stagefright/recordvideo.cpp
index a63b9b9984..01a178e6a0 100644
--- a/cmds/stagefright/recordvideo.cpp
+++ b/cmds/stagefright/recordvideo.cpp
@@ -14,8 +14,6 @@
  * limitations under the License.
  */
 
-#include "SineSource.h"
-
 #include <inttypes.h>
 #include <sys/types.h>
 #include <sys/stat.h>
@@ -25,8 +23,8 @@
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/ALooper.h>
 #include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/AudioPlayer.h>
 #include <media/stagefright/MediaBufferGroup.h>
+#include <media/stagefright/MediaBuffer.h>
 #include <media/stagefright/MediaCodecSource.h>
 #include <media/stagefright/MediaDefs.h>
 #include <media/stagefright/MetaData.h>
diff --git a/cmds/stagefright/stagefright.cpp b/cmds/stagefright/stagefright.cpp
index d52541d2df..a1ee90483f 100644
--- a/cmds/stagefright/stagefright.cpp
+++ b/cmds/stagefright/stagefright.cpp
@@ -40,7 +40,6 @@
 #include <media/stagefright/foundation/ALooper.h>
 #include <media/stagefright/foundation/AMessage.h>
 #include <media/stagefright/foundation/AUtils.h>
-#include <media/stagefright/AudioPlayer.h>
 #include <media/stagefright/JPEGSource.h>
 #include <media/stagefright/InterfaceUtils.h>
 #include <media/stagefright/MediaCodec.h>
@@ -67,6 +66,8 @@
 
 #include <android/hardware/media/omx/1.0/IOmx.h>
 
+#include "AudioPlayer.h"
+
 using namespace android;
 
 static long gNumRepetitions;