Squashed commit of the following:

commit 4abf16bb04dc9695fedf4007a84f903074312ccd
Author: Andreas Huber <andih@google.com>
Date:   Tue Jul 20 09:21:17 2010 -0700

    Support a single format change at the beginning of audio playback. This way the AAC+ decoder may change its output format from what is originally encoded in the audio stream and we'll still play it back correctly.

    Change-Id: Icc790122744745e9a88099788d4818ca1e265a82
    related-to-bug: 2826841

commit 09c74da63e6ad5cb5dafb70f62696d75d2978967
Author: James Dong <jdong@google.com>
Date:   Sun Jul 18 17:57:01 2010 -0700

    Fix MPEG4Extractor to extract sampling frequency correctly when SBR is enabled.

    Change-Id: I883c81dad3ea465e71cb5590e89d763671a90ff8

commit f672bf2a782dc7d5fb6325d611a7fe17045dfe9a
Author: James Dong <jdong@google.com>
Date:   Thu Jul 8 20:56:13 2010 -0700

    Enable the support for decoding audio with AAC+ and eAAC+ features

    bug - 282684

    Change-Id: I73c8377af3cc4edd3ee7cea86dc3b1c369fbd78b

Change-Id: I012f1179e933b6d1345d2368f357576c722485f7

include/media/stagefright/AudioPlayer.h

@@ -86,6 +86,10 @@ private:
 
     bool mStarted;
 
+    bool mIsFirstBuffer;
+    status_t mFirstBufferResult;
+    MediaBuffer *mFirstBuffer;
+
     sp<MediaPlayerBase::AudioSink> mAudioSink;
 
     static void AudioCallback(int event, void *user, void *info);

media/libstagefright/AudioPlayer.cpp

@@ -23,6 +23,7 @@
 #include <media/stagefright/AudioPlayer.h>
 #include <media/stagefright/MediaDebug.h>
 #include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MediaErrors.h>
 #include <media/stagefright/MediaSource.h>
 #include <media/stagefright/MetaData.h>
 
@@ -41,6 +42,9 @@ AudioPlayer::AudioPlayer(const sp<MediaPlayerBase::AudioSink> &audioSink)
       mReachedEOS(false),
       mFinalStatus(OK),
       mStarted(false),
+      mIsFirstBuffer(false),
+      mFirstBufferResult(OK),
+      mFirstBuffer(NULL),
       mAudioSink(audioSink) {
 }
 
@@ -68,6 +72,24 @@ status_t AudioPlayer::start(bool sourceAlreadyStarted) {
         }
     }
 
+    // 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);
+
+    mFirstBufferResult = mSource->read(&mFirstBuffer);
+    if (mFirstBufferResult == INFO_FORMAT_CHANGED) {
+        LOGV("INFO_FORMAT_CHANGED!!!");
+
+        CHECK(mFirstBuffer == NULL);
+        mFirstBufferResult = OK;
+        mIsFirstBuffer = false;
+    } else {
+        mIsFirstBuffer = true;
+    }
+
     sp<MetaData> format = mSource->getFormat();
     const char *mime;
     bool success = format->findCString(kKeyMIMEType, &mime);
@@ -87,7 +109,14 @@ status_t AudioPlayer::start(bool sourceAlreadyStarted) {
                 DEFAULT_AUDIOSINK_BUFFERCOUNT,
                 &AudioPlayer::AudioSinkCallback, this);
         if (err != OK) {
-            mSource->stop();
+            if (mFirstBuffer != NULL) {
+                mFirstBuffer->release();
+                mFirstBuffer = NULL;
+            }
+
+            if (!sourceAlreadyStarted) {
+                mSource->stop();
+            }
 
             return err;
         }
@@ -108,7 +137,14 @@ status_t AudioPlayer::start(bool sourceAlreadyStarted) {
             delete mAudioTrack;
             mAudioTrack = NULL;
 
-            mSource->stop();
+            if (mFirstBuffer != NULL) {
+                mFirstBuffer->release();
+                mFirstBuffer = NULL;
+            }
+
+            if (!sourceAlreadyStarted) {
+                mSource->stop();
+            }
 
             return err;
         }
@@ -159,6 +195,12 @@ void AudioPlayer::stop() {
 
     // 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) {
         LOGV("AudioPlayer releasing input buffer.");
 
@@ -243,6 +285,14 @@ size_t AudioPlayer::fillBuffer(void *data, size_t size) {
             Mutex::Autolock autoLock(mLock);
 
             if (mSeeking) {
+                if (mIsFirstBuffer) {
+                    if (mFirstBuffer != NULL) {
+                        mFirstBuffer->release();
+                        mFirstBuffer = NULL;
+                    }
+                    mIsFirstBuffer = false;
+                }
+
                 options.setSeekTo(mSeekTimeUs);
 
                 if (mInputBuffer != NULL) {
@@ -255,7 +305,17 @@ size_t AudioPlayer::fillBuffer(void *data, size_t size) {
         }
 
         if (mInputBuffer == NULL) {
-            status_t err = mSource->read(&mInputBuffer, &options);
+            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));

media/libstagefright/codecs/aacdec/AACDecoder.cpp

@@ -15,6 +15,7 @@
  */
 
 #include "AACDecoder.h"
+#define LOG_TAG "AACDecoder"
 
 #include "../../include/ESDS.h"
 
@@ -36,26 +37,33 @@ AACDecoder::AACDecoder(const sp<MediaSource> &source)
       mAnchorTimeUs(0),
       mNumSamplesOutput(0),
       mInputBuffer(NULL) {
-}
 
-AACDecoder::~AACDecoder() {
+    sp<MetaData> srcFormat = mSource->getFormat();
-    if (mStarted) {
+
-        stop();
+    int32_t sampleRate;
+    CHECK(srcFormat->findInt32(kKeySampleRate, &sampleRate));
+
+    mMeta = new MetaData;
+    mMeta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_RAW);
+
+    // We'll always output stereo, regardless of how many channels are
+    // present in the input due to decoder limitations.
+    mMeta->setInt32(kKeyChannelCount, 2);
+    mMeta->setInt32(kKeySampleRate, sampleRate);
+
+    int64_t durationUs;
+    if (srcFormat->findInt64(kKeyDuration, &durationUs)) {
+        mMeta->setInt64(kKeyDuration, durationUs);
     }
+    mMeta->setCString(kKeyDecoderComponent, "AACDecoder");
 
-    delete mConfig;
+    mInitCheck = initCheck();
-    mConfig = NULL;
 }
 
-status_t AACDecoder::start(MetaData *params) {
+status_t AACDecoder::initCheck() {
-    CHECK(!mStarted);
+    memset(mConfig, 0, sizeof(tPVMP4AudioDecoderExternal));
-
-    mBufferGroup = new MediaBufferGroup;
-    mBufferGroup->add_buffer(new MediaBuffer(2048 * 2));
-
     mConfig->outputFormat = OUTPUTFORMAT_16PCM_INTERLEAVED;
-    mConfig->aacPlusUpsamplingFactor = 0;
+    mConfig->aacPlusEnabled = 1;
-    mConfig->aacPlusEnabled = false;
 
     // The software decoder doesn't properly support mono output on
     // AACplus files. Always output stereo.
@@ -64,8 +72,11 @@ status_t AACDecoder::start(MetaData *params) {
     UInt32 memRequirements = PVMP4AudioDecoderGetMemRequirements();
     mDecoderBuf = malloc(memRequirements);
 
-    CHECK_EQ(PVMP4AudioDecoderInitLibrary(mConfig, mDecoderBuf),
+    status_t err = PVMP4AudioDecoderInitLibrary(mConfig, mDecoderBuf);
-             MP4AUDEC_SUCCESS);
+    if (err != MP4AUDEC_SUCCESS) {
+        LOGE("Failed to initialize MP4 audio decoder");
+        return UNKNOWN_ERROR;
+    }
 
     uint32_t type;
     const void *data;
@@ -83,18 +94,29 @@ status_t AACDecoder::start(MetaData *params) {
         mConfig->pInputBuffer = (UChar *)codec_specific_data;
         mConfig->inputBufferCurrentLength = codec_specific_data_size;
         mConfig->inputBufferMaxLength = 0;
-        mConfig->inputBufferUsedLength = 0;
-        mConfig->remainderBits = 0;
-
-        mConfig->pOutputBuffer = NULL;
-        mConfig->pOutputBuffer_plus = NULL;
-        mConfig->repositionFlag = false;
 
         if (PVMP4AudioDecoderConfig(mConfig, mDecoderBuf)
                 != MP4AUDEC_SUCCESS) {
             return ERROR_UNSUPPORTED;
         }
     }
+    return OK;
+}
+
+AACDecoder::~AACDecoder() {
+    if (mStarted) {
+        stop();
+    }
+
+    delete mConfig;
+    mConfig = NULL;
+}
+
+status_t AACDecoder::start(MetaData *params) {
+    CHECK(!mStarted);
+
+    mBufferGroup = new MediaBufferGroup;
+    mBufferGroup->add_buffer(new MediaBuffer(4096 * 2));
 
     mSource->start();
 
@@ -127,28 +149,7 @@ status_t AACDecoder::stop() {
 }
 
 sp<MetaData> AACDecoder::getFormat() {
-    sp<MetaData> srcFormat = mSource->getFormat();
+    return mMeta;
-
-    int32_t sampleRate;
-    CHECK(srcFormat->findInt32(kKeySampleRate, &sampleRate));
-
-    sp<MetaData> meta = new MetaData;
-    meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_RAW);
-
-    // We'll always output stereo, regardless of how many channels are
-    // present in the input due to decoder limitations.
-    meta->setInt32(kKeyChannelCount, 2);
-
-    meta->setInt32(kKeySampleRate, sampleRate);
-
-    int64_t durationUs;
-    if (srcFormat->findInt64(kKeyDuration, &durationUs)) {
-        meta->setInt64(kKeyDuration, durationUs);
-    }
-
-    meta->setCString(kKeyDecoderComponent, "AACDecoder");
-
-    return meta;
 }
 
 status_t AACDecoder::read(
@@ -200,13 +201,32 @@ status_t AACDecoder::read(
     mConfig->remainderBits = 0;
 
     mConfig->pOutputBuffer = static_cast<Int16 *>(buffer->data());
-    mConfig->pOutputBuffer_plus = NULL;
+    mConfig->pOutputBuffer_plus = &mConfig->pOutputBuffer[2048];
     mConfig->repositionFlag = false;
 
     Int decoderErr = PVMP4AudioDecodeFrame(mConfig, mDecoderBuf);
 
+    // Check on the sampling rate to see whether it is changed.
+    int32_t sampleRate;
+    CHECK(mMeta->findInt32(kKeySampleRate, &sampleRate));
+    if (mConfig->samplingRate != sampleRate) {
+        mMeta->setInt32(kKeySampleRate, mConfig->samplingRate);
+        LOGW("Sample rate was %d, but now is %d",
+                sampleRate, mConfig->samplingRate);
+        buffer->release();
+        mInputBuffer->release();
+        mInputBuffer = NULL;
+        return INFO_FORMAT_CHANGED;
+    }
+
     size_t numOutBytes =
         mConfig->frameLength * sizeof(int16_t) * mConfig->desiredChannels;
+    if (mConfig->aacPlusUpsamplingFactor == 2) {
+        if (mConfig->desiredChannels == 1) {
+            memcpy(&mConfig->pOutputBuffer[1024], &mConfig->pOutputBuffer[2048], numOutBytes * 2);
+        }
+        numOutBytes *= 2;
+    }
 
     if (decoderErr != MP4AUDEC_SUCCESS) {
         LOGW("AAC decoder returned error %d, substituting silence", decoderErr);

media/libstagefright/include/AACDecoder.h

@@ -25,6 +25,7 @@ struct tPVMP4AudioDecoderExternal;
 namespace android {
 
 struct MediaBufferGroup;
+struct MetaData;
 
 struct AACDecoder : public MediaSource {
     AACDecoder(const sp<MediaSource> &source);
@@ -41,6 +42,7 @@ protected:
     virtual ~AACDecoder();
 
 private:
+    sp<MetaData>    mMeta;
     sp<MediaSource> mSource;
     bool mStarted;
 
@@ -50,9 +52,11 @@ private:
     void *mDecoderBuf;
     int64_t mAnchorTimeUs;
     int64_t mNumSamplesOutput;
+    status_t mInitCheck;
 
     MediaBuffer *mInputBuffer;
 
+    status_t initCheck();
     AACDecoder(const AACDecoder &);
     AACDecoder &operator=(const AACDecoder &);
 };