* limitations under the License.
*/
-//#define LOG_NDEBUG 0
#define LOG_TAG "SoftFFmpegAudio"
#include <utils/Log.h>
#include <cutils/properties.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/hexdump.h>
+#include <media/stagefright/ACodec.h>
#include <media/stagefright/MediaDefs.h>
-#include <media/stagefright/OMXCodec.h>
#include <OMX_AudioExt.h>
#include <OMX_IndexExt.h>
#define DEBUG_PKT 0
#define DEBUG_FRM 0
+#define DEBUG_EXTRADATA 0
namespace android {
template<class T>
static void InitOMXParams(T *params) {
+ memset(params, 0, sizeof(T));
params->nSize = sizeof(T);
params->nVersion.s.nVersionMajor = 1;
params->nVersion.s.nVersionMinor = 0;
int fast = 0;
avctx->workaround_bugs = 1;
- avctx->lowres = 0;
- if(avctx->lowres > codec->max_lowres){
- ALOGW("The maximum value for lowres supported by the decoder is %d",
- codec->max_lowres);
- avctx->lowres= codec->max_lowres;
- }
avctx->idct_algo = 0;
avctx->skip_frame = AVDISCARD_DEFAULT;
avctx->skip_idct = AVDISCARD_DEFAULT;
avctx->skip_loop_filter = AVDISCARD_DEFAULT;
avctx->error_concealment = 3;
- avctx->flags |= CODEC_FLAG_BITEXACT;
+ avctx->flags |= AV_CODEC_FLAG_BITEXACT;
- if(avctx->lowres) avctx->flags |= CODEC_FLAG_EMU_EDGE;
- if (fast) avctx->flags2 |= CODEC_FLAG2_FAST;
- if(codec->capabilities & CODEC_CAP_DR1)
+ if (fast) avctx->flags2 |= AV_CODEC_FLAG2_FAST;
+#ifdef CODEC_FLAG_EMU_EDGE
+ if (codec->capabilities & AV_CODEC_CAP_DR1)
avctx->flags |= CODEC_FLAG_EMU_EDGE;
+#endif
}
bool SoftFFmpegAudio::isConfigured() {
mCtx = NULL;
}
if (mFrame) {
- av_freep(&mFrame);
+ av_frame_free(&mFrame);
mFrame = NULL;
}
+#ifdef LIBAV_CONFIG_H
+#else
if (mSwrCtx) {
swr_free(&mSwrCtx);
mSwrCtx = NULL;
}
+#endif
}
OMX_ERRORTYPE SoftFFmpegAudio::internalGetParameter(
profile->eEndian = OMX_EndianBig;
profile->bInterleaved = OMX_TRUE;
profile->ePCMMode = OMX_AUDIO_PCMModeLinear;
+ profile->nBitPerSample = 32;
+ profile->eNumData = OMX_NumericalDataFloat;
if (isConfigured()) {
- AVSampleFormat packed = av_get_packed_sample_fmt(mAudioTgtFmt);
- if (packed == AV_SAMPLE_FMT_U8)
- profile->nBitPerSample = 8;
- else if (packed == AV_SAMPLE_FMT_S16)
- profile->nBitPerSample = 16;
- else if (packed == AV_SAMPLE_FMT_S32)
- profile->nBitPerSample = 24;
- else
- profile->nBitPerSample = av_get_bytes_per_sample(mAudioTgtFmt) * 8;
- } else {
- profile->nBitPerSample = 32;
+ switch (av_get_packed_sample_fmt(mAudioTgtFmt)) {
+ case AV_SAMPLE_FMT_U8:
+ profile->nBitPerSample = 8;
+ profile->eNumData = OMX_NumericalDataUnsigned;
+ break;
+ case AV_SAMPLE_FMT_S16:
+ profile->nBitPerSample = 16;
+ profile->eNumData = OMX_NumericalDataSigned;
+ break;
+ case AV_SAMPLE_FMT_S32:
+ profile->nBitPerSample = 32;
+ profile->eNumData = OMX_NumericalDataSigned;
+ break;
+ default:
+ profile->nBitPerSample = 32;
+ profile->eNumData = OMX_NumericalDataFloat;
+ break;
+ }
}
- if (getOMXChannelMapping(mAudioTgtChannels, profile->eChannelMapping) != OK) {
+ if (ACodec::getOMXChannelMapping(mAudioTgtChannels, profile->eChannelMapping) != OK) {
return OMX_ErrorNone;
}
profile->nSamplingRate = mAudioTgtFreq;
//mCtx has been updated(adjustAudioParams)!
- ALOGV("get pcm params, nChannels:%u, nSamplingRate:%u, nBitsPerSample:%u",
+ ALOGV("get pcm params, nChannels:%u, nSamplingRate:%u, nBitPerSample:%u",
profile->nChannels, profile->nSamplingRate, profile->nBitPerSample);
return OMX_ErrorNone;
profile->nChannels = mCtx->channels;
profile->nSampleRate = mCtx->sample_rate;
-
+ profile->nCompressionLevel = mCtx->bits_per_raw_sample;
return OMX_ErrorNone;
}
return OMX_ErrorNone;
}
+ case OMX_IndexParamAudioAlac:
+ {
+ OMX_AUDIO_PARAM_ALACTYPE *profile =
+ (OMX_AUDIO_PARAM_ALACTYPE *)params;
+
+ if (profile->nPortIndex != kInputPortIndex) {
+ return OMX_ErrorUndefined;
+ }
+
+ profile->nChannels = mCtx->channels;
+ profile->nSamplingRate = mCtx->sample_rate;
+
+ profile->nBitsPerSample = mCtx->bits_per_coded_sample;
+
+ return OMX_ErrorNone;
+ }
+
case OMX_IndexParamAudioApe:
{
OMX_AUDIO_PARAM_APETYPE *profile =
return OMX_ErrorUndefined;
}
- if (profile->nBitPerSample == 24) {
- mAudioTgtFmt = AV_SAMPLE_FMT_S32;
- } else if (profile->nBitPerSample == 32) {
- mAudioTgtFmt = AV_SAMPLE_FMT_FLT;
- } else if (profile->nBitPerSample == 8) {
- mAudioTgtFmt = AV_SAMPLE_FMT_U8;
- } else {
- mAudioTgtFmt = AV_SAMPLE_FMT_S16;
+ switch (profile->nBitPerSample) {
+ case 8:
+ mAudioTgtFmt = AV_SAMPLE_FMT_U8;
+ break;
+ case 16:
+ mAudioTgtFmt = AV_SAMPLE_FMT_S16;
+ break;
+ case 24:
+ mAudioTgtFmt = AV_SAMPLE_FMT_S32;
+ break;
+ case 32:
+ if (profile->eNumData == OMX_NumericalDataFloat) {
+ mAudioTgtFmt = AV_SAMPLE_FMT_FLT;
+ break;
+ }
+ mAudioTgtFmt = AV_SAMPLE_FMT_S32;
+ break;
+ default:
+ ALOGE("Unknown PCM encoding, assuming floating point");
+ mAudioTgtFmt = AV_SAMPLE_FMT_FLT;
}
mAudioTgtFreq = profile->nSamplingRate;
mAudioTgtChannels = profile->nChannels;
ALOGV("set OMX_IndexParamAudioPcm, nChannels:%u, "
- "nSampleRate:%u, nBitsPerSample:%u",
+ "nSampleRate:%u, nBitPerSample:%u",
profile->nChannels, profile->nSamplingRate,
profile->nBitPerSample);
return OMX_ErrorNone;
}
+ case OMX_IndexParamAudioAlac:
+ {
+ OMX_AUDIO_PARAM_ALACTYPE *profile =
+ (OMX_AUDIO_PARAM_ALACTYPE *)params;
+
+ if (profile->nPortIndex != kInputPortIndex) {
+ return OMX_ErrorUndefined;
+ }
+
+ mCtx->channels = profile->nChannels;
+ mCtx->sample_rate = profile->nSamplingRate;
+ mCtx->bits_per_coded_sample = profile->nBitsPerSample;
+
+ adjustAudioParams();
+
+ ALOGV("set OMX_IndexParamAudioAlac, nChannels:%u, "
+ "nSampleRate:%u, nBitsPerSample:%u",
+ profile->nChannels, profile->nSamplingRate,
+ profile->nBitsPerSample);
+
+ return OMX_ErrorNone;
+ }
case OMX_IndexParamAudioApe:
{
BufferInfo *inInfo = *inQueue.begin();
OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+#if DEBUG_EXTRADATA
ALOGI("got extradata, ignore: %d, size: %u",
mIgnoreExtradata, inHeader->nFilledLen);
hexdump(inHeader->pBuffer + inHeader->nOffset, inHeader->nFilledLen);
+#endif
if (mIgnoreExtradata) {
ALOGI("got extradata, size: %u, but ignore it", inHeader->nFilledLen);
int orig_extradata_size = mCtx->extradata_size;
mCtx->extradata_size += inHeader->nFilledLen;
mCtx->extradata = (uint8_t *)av_realloc(mCtx->extradata,
- mCtx->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE);
+ mCtx->extradata_size + AV_INPUT_BUFFER_PADDING_SIZE);
if (!mCtx->extradata) {
ALOGE("ffmpeg audio decoder failed to alloc extradata memory.");
return ERR_OOM;
memcpy(mCtx->extradata + orig_extradata_size,
inHeader->pBuffer + inHeader->nOffset, inHeader->nFilledLen);
- memset(mCtx->extradata + mCtx->extradata_size, 0, FF_INPUT_BUFFER_PADDING_SIZE);
+ memset(mCtx->extradata + mCtx->extradata_size, 0, AV_INPUT_BUFFER_PADDING_SIZE);
}
}
}
}
deinitVorbisHdr();
}
+#if DEBUG_EXTRADATA
ALOGI("extradata is ready, size: %d", mCtx->extradata_size);
hexdump(mCtx->extradata, mCtx->extradata_size);
+#endif
mExtradataReady = true;
}
}
if (mEOSStatus == INPUT_EOS_SEEN && (!inHeader || inHeader->nFilledLen == 0)
- && !(mCtx->codec->capabilities & CODEC_CAP_DELAY)) {
+ && !(mCtx->codec->capabilities & AV_CODEC_CAP_DELAY)) {
return ERR_FLUSHED;
}
AVPacket pkt;
initPacket(&pkt, inHeader);
- av_frame_unref(mFrame);
len = avcodec_decode_audio4(mCtx, mFrame, &gotFrm, &pkt);
+ av_packet_unref(&pkt);
+
//a negative error code is returned if an error occurred during decoding
if (len < 0) {
ALOGW("ffmpeg audio decoder err, we skip the frame and play silence instead");
ALOGI("ffmpeg audio decoder failed to get frame.");
#endif
//stop sending empty packets if the decoder is finished
- if (is_flush && mCtx->codec->capabilities & CODEC_CAP_DELAY) {
+ if (is_flush && mCtx->codec->capabilities & AV_CODEC_CAP_DELAY) {
ALOGI("ffmpeg audio decoder failed to get more frames when flush.");
ret = ERR_FLUSHED;
} else {
} else {
inputBufferUsedLength = len;
}
-
- CHECK_GE(inHeader->nFilledLen, inputBufferUsedLength);
- inHeader->nOffset += inputBufferUsedLength;
- inHeader->nFilledLen -= inputBufferUsedLength;
mInputBufferSize -= inputBufferUsedLength;
- if (inHeader->nFilledLen == 0) {
- CHECK_EQ(mInputBufferSize, 0);
- inQueue.erase(inQueue.begin());
- inInfo->mOwnedByUs = false;
- notifyEmptyBufferDone(inHeader);
+ if (inHeader != NULL) {
+ CHECK_GE(inHeader->nFilledLen, inputBufferUsedLength);
+ inHeader->nOffset += inputBufferUsedLength;
+ inHeader->nFilledLen -= inputBufferUsedLength;
+
+ if (inHeader->nFilledLen == 0) {
+ CHECK_EQ(mInputBufferSize, 0);
+ inQueue.erase(inQueue.begin());
+ inInfo->mOwnedByUs = false;
+ notifyEmptyBufferDone(inHeader);
+ }
}
}
return ret;
}
+#ifdef LIBAV_CONFIG_H
+#define av_frame_get_channels(f) av_get_channel_layout_nb_channels(f->channel_layout)
+#endif
+
int32_t SoftFFmpegAudio::resampleAudio() {
int channels = 0;
int64_t channelLayout = 0;
|| mAudioSrcFmt != mAudioTgtFmt
|| mAudioSrcChannelLayout != mAudioTgtChannelLayout
|| mAudioSrcFreq != mAudioTgtFreq))) {
+#ifdef LIBAV_CONFIG_H
+ if (!mSwrCtx) {
+#else
if (mSwrCtx) {
swr_free(&mSwrCtx);
}
channelLayout, (enum AVSampleFormat)mFrame->format, mFrame->sample_rate,
0, NULL);
if (!mSwrCtx || swr_init(mSwrCtx) < 0) {
+#endif
ALOGE("Cannot create sample rate converter for conversion "
"of %d Hz %s %d channels to %d Hz %s %d channels!",
mFrame->sample_rate,
}
if (mSwrCtx) {
+#ifdef LIBAV_CONFIG_H
+#else
const uint8_t **in = (const uint8_t **)mFrame->extended_data;
uint8_t *out[] = {mAudioBuffer};
int out_count = sizeof(mAudioBuffer) / mAudioTgtChannels / av_get_bytes_per_sample(mAudioTgtFmt);
mAudioTgtChannels,
av_get_sample_fmt_name(mAudioTgtFmt));
#endif
+#endif
} else {
mResampledData = mFrame->data[0];
mResampledDataSize = dataSize;
sAudioClock = (int64_t*) malloc(sizeof(int64_t));
*sAudioClock = 0;
}
- ALOGV("getAudioClock: %lld", *sAudioClock);
+ ALOGV("getAudioClock: %" PRId64, *sAudioClock);
return *sAudioClock;
}
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