static
void put_tool_tag(m4af_ctx_t *m4af, const aacenc_param_ex_t *params,
HANDLE_AACENCODER encoder)
{
char tool_info[256];
char *p = tool_info;
LIB_INFO *lib_info = 0;
p += sprintf(p, PROGNAME " %s, ", fdkaac_version);
lib_info = calloc(FDK_MODULE_LAST, sizeof(LIB_INFO));
if (aacEncGetLibInfo(lib_info) == AACENC_OK) {
int i;
for (i = 0; i < FDK_MODULE_LAST; ++i)
if (lib_info[i].module_id == FDK_AACENC)
break;
p += sprintf(p, "libfdk-aac %s, ", lib_info[i].versionStr);
}
free(lib_info);
if (params->bitrate_mode)
sprintf(p, "VBR mode %d", params->bitrate_mode);
else
sprintf(p, "CBR %dkbps",
aacEncoder_GetParam(encoder, AACENC_BITRATE) / 1000);
m4af_add_itmf_string_tag(m4af, M4AF_TAG_TOOL, tool_info);
}
int aac_encode_frame(HANDLE_AACENCODER encoder,
const pcm_sample_description_t *format,
const int16_t *input, unsigned iframes,
aacenc_frame_t *output)
{
uint32_t ilen = iframes * format->channels_per_frame;
AACENC_BufDesc ibdesc = { 0 }, obdesc = { 0 };
AACENC_InArgs iargs = { 0 };
AACENC_OutArgs oargs = { 0 };
void *ibufs[] = { (void*)input };
void *obufs[1];
INT ibuf_ids[] = { IN_AUDIO_DATA };
INT obuf_ids[] = { OUT_BITSTREAM_DATA };
INT ibuf_sizes[] = { ilen * sizeof(int16_t) };
INT obuf_sizes[1];
INT ibuf_el_sizes[] = { sizeof(int16_t) };
INT obuf_el_sizes[] = { 1 };
AACENC_ERROR err;
unsigned channel_mode, obytes;
channel_mode = aacEncoder_GetParam(encoder, AACENC_CHANNELMODE);
obytes = 6144 / 8 * channel_mode;
if (!output->data || output->capacity < obytes) {
uint8_t *p = realloc(output->data, obytes);
if (!p) return -1;
output->capacity = obytes;
output->data = p;
}
obufs[0] = output->data;
obuf_sizes[0] = obytes;
iargs.numInSamples = ilen ? ilen : -1; /* -1 for signaling EOF */
ibdesc.numBufs = 1;
ibdesc.bufs = ibufs;
ibdesc.bufferIdentifiers = ibuf_ids;
ibdesc.bufSizes = ibuf_sizes;
ibdesc.bufElSizes = ibuf_el_sizes;
obdesc.numBufs = 1;
obdesc.bufs = obufs;
obdesc.bufferIdentifiers = obuf_ids;
obdesc.bufSizes = obuf_sizes;
obdesc.bufElSizes = obuf_el_sizes;
err = aacEncEncode(encoder, &ibdesc, &obdesc, &iargs, &oargs);
if (err != AACENC_ENCODE_EOF && err != AACENC_OK) {
fprintf(stderr, "ERROR: aacEncEncode() failed\n");
return -1;
}
output->size = oargs.numOutBytes;
return oargs.numInSamples / format->channels_per_frame;
}
static
void put_tool_tag(m4af_ctx_t *m4af, const aacenc_param_ex_t *params,
HANDLE_AACENCODER encoder)
{
char tool_info[256];
char *p = tool_info;
LIB_INFO lib_info;
p += sprintf(p, PROGNAME " %s, ", fdkaac_version);
aacenc_get_lib_info(&lib_info);
p += sprintf(p, "libfdk-aac %s, ", lib_info.versionStr);
if (params->bitrate_mode)
sprintf(p, "VBR mode %d", params->bitrate_mode);
else
sprintf(p, "CBR %dkbps",
aacEncoder_GetParam(encoder, AACENC_BITRATE) / 1000);
m4af_add_itmf_string_tag(m4af, M4AF_TAG_TOOL, tool_info);
}
void SoftAACEncoder2::onQueueFilled(OMX_U32 /* portIndex */) {
if (mSignalledError) {
return;
}
List<BufferInfo *> &inQueue = getPortQueue(0);
List<BufferInfo *> &outQueue = getPortQueue(1);
if (!mSentCodecSpecificData) {
// The very first thing we want to output is the codec specific
// data. It does not require any input data but we will need an
// output buffer to store it in.
if (outQueue.empty()) {
return;
}
if (AACENC_OK != aacEncEncode(mAACEncoder, NULL, NULL, NULL, NULL)) {
ALOGE("Unable to initialize encoder for profile / sample-rate / bit-rate / channels");
notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
mSignalledError = true;
return;
}
OMX_U32 actualBitRate = aacEncoder_GetParam(mAACEncoder, AACENC_BITRATE);
if (mBitRate != actualBitRate) {
ALOGW("Requested bitrate %u unsupported, using %u", mBitRate, actualBitRate);
}
AACENC_InfoStruct encInfo;
if (AACENC_OK != aacEncInfo(mAACEncoder, &encInfo)) {
ALOGE("Failed to get AAC encoder info");
notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
mSignalledError = true;
return;
}
BufferInfo *outInfo = *outQueue.begin();
OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
if (outHeader->nOffset + encInfo.confSize > outHeader->nAllocLen) {
ALOGE("b/34617444");
android_errorWriteLog(0x534e4554,"34617444");
notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
mSignalledError = true;
return;
}
outHeader->nFilledLen = encInfo.confSize;
outHeader->nFlags = OMX_BUFFERFLAG_CODECCONFIG;
uint8_t *out = outHeader->pBuffer + outHeader->nOffset;
memcpy(out, encInfo.confBuf, encInfo.confSize);
outQueue.erase(outQueue.begin());
outInfo->mOwnedByUs = false;
notifyFillBufferDone(outHeader);
mSentCodecSpecificData = true;
}
size_t numBytesPerInputFrame =
mNumChannels * kNumSamplesPerFrame * sizeof(int16_t);
// Limit input size so we only get one ELD frame
if (mAACProfile == OMX_AUDIO_AACObjectELD && numBytesPerInputFrame > 512) {
numBytesPerInputFrame = 512;
}
for (;;) {
// We do the following until we run out of buffers.
while (mInputSize < numBytesPerInputFrame) {
// As long as there's still input data to be read we
// will drain "kNumSamplesPerFrame * mNumChannels" samples
// into the "mInputFrame" buffer and then encode those
// as a unit into an output buffer.
if (mSawInputEOS || inQueue.empty()) {
return;
}
BufferInfo *inInfo = *inQueue.begin();
OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
const void *inData = inHeader->pBuffer + inHeader->nOffset;
size_t copy = numBytesPerInputFrame - mInputSize;
if (copy > inHeader->nFilledLen) {
copy = inHeader->nFilledLen;
}
if (mInputFrame == NULL) {
mInputFrame = new int16_t[numBytesPerInputFrame / sizeof(int16_t)];
mAllocatedFrameSize = numBytesPerInputFrame;
} else if (mAllocatedFrameSize != numBytesPerInputFrame) {
ALOGE("b/34621073: changed size from %d to %d",
(int)mAllocatedFrameSize, (int)numBytesPerInputFrame);
android_errorWriteLog(0x534e4554,"34621073");
delete mInputFrame;
mInputFrame = new int16_t[numBytesPerInputFrame / sizeof(int16_t)];
mAllocatedFrameSize = numBytesPerInputFrame;
}
if (mInputSize == 0) {
mInputTimeUs = inHeader->nTimeStamp;
}
memcpy((uint8_t *)mInputFrame + mInputSize, inData, copy);
mInputSize += copy;
inHeader->nOffset += copy;
inHeader->nFilledLen -= copy;
// "Time" on the input buffer has in effect advanced by the
// number of audio frames we just advanced nOffset by.
inHeader->nTimeStamp +=
(copy * 1000000ll / mSampleRate)
/ (mNumChannels * sizeof(int16_t));
if (inHeader->nFilledLen == 0) {
if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
mSawInputEOS = true;
// Pad any remaining data with zeroes.
memset((uint8_t *)mInputFrame + mInputSize,
0,
numBytesPerInputFrame - mInputSize);
mInputSize = numBytesPerInputFrame;
}
inQueue.erase(inQueue.begin());
inInfo->mOwnedByUs = false;
notifyEmptyBufferDone(inHeader);
inData = NULL;
inHeader = NULL;
inInfo = NULL;
}
}
// At this point we have all the input data necessary to encode
// a single frame, all we need is an output buffer to store the result
// in.
if (outQueue.empty()) {
return;
}
BufferInfo *outInfo = *outQueue.begin();
OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
uint8_t *outPtr = (uint8_t *)outHeader->pBuffer + outHeader->nOffset;
size_t outAvailable = outHeader->nAllocLen - outHeader->nOffset;
AACENC_InArgs inargs;
AACENC_OutArgs outargs;
memset(&inargs, 0, sizeof(inargs));
memset(&outargs, 0, sizeof(outargs));
inargs.numInSamples = numBytesPerInputFrame / sizeof(int16_t);
void* inBuffer[] = { (unsigned char *)mInputFrame };
INT inBufferIds[] = { IN_AUDIO_DATA };
INT inBufferSize[] = { (INT)numBytesPerInputFrame };
INT inBufferElSize[] = { sizeof(int16_t) };
AACENC_BufDesc inBufDesc;
inBufDesc.numBufs = sizeof(inBuffer) / sizeof(void*);
inBufDesc.bufs = (void**)&inBuffer;
inBufDesc.bufferIdentifiers = inBufferIds;
inBufDesc.bufSizes = inBufferSize;
inBufDesc.bufElSizes = inBufferElSize;
void* outBuffer[] = { outPtr };
INT outBufferIds[] = { OUT_BITSTREAM_DATA };
INT outBufferSize[] = { 0 };
INT outBufferElSize[] = { sizeof(UCHAR) };
AACENC_BufDesc outBufDesc;
outBufDesc.numBufs = sizeof(outBuffer) / sizeof(void*);
outBufDesc.bufs = (void**)&outBuffer;
outBufDesc.bufferIdentifiers = outBufferIds;
outBufDesc.bufSizes = outBufferSize;
outBufDesc.bufElSizes = outBufferElSize;
// Encode the mInputFrame, which is treated as a modulo buffer
AACENC_ERROR encoderErr = AACENC_OK;
size_t nOutputBytes = 0;
do {
memset(&outargs, 0, sizeof(outargs));
outBuffer[0] = outPtr;
outBufferSize[0] = outAvailable - nOutputBytes;
encoderErr = aacEncEncode(mAACEncoder,
&inBufDesc,
&outBufDesc,
&inargs,
&outargs);
if (encoderErr == AACENC_OK) {
outPtr += outargs.numOutBytes;
nOutputBytes += outargs.numOutBytes;
if (outargs.numInSamples > 0) {
int numRemainingSamples = inargs.numInSamples - outargs.numInSamples;
if (numRemainingSamples > 0) {
memmove(mInputFrame,
&mInputFrame[outargs.numInSamples],
sizeof(int16_t) * numRemainingSamples);
}
inargs.numInSamples -= outargs.numInSamples;
}
}
} while (encoderErr == AACENC_OK && inargs.numInSamples > 0);
outHeader->nFilledLen = nOutputBytes;
outHeader->nFlags = OMX_BUFFERFLAG_ENDOFFRAME;
if (mSawInputEOS) {
// We also tag this output buffer with EOS if it corresponds
// to the final input buffer.
outHeader->nFlags = OMX_BUFFERFLAG_EOS;
}
outHeader->nTimeStamp = mInputTimeUs;
#if 0
ALOGI("sending %d bytes of data (time = %lld us, flags = 0x%08lx)",
nOutputBytes, mInputTimeUs, outHeader->nFlags);
hexdump(outHeader->pBuffer + outHeader->nOffset, outHeader->nFilledLen);
#endif
outQueue.erase(outQueue.begin());
outInfo->mOwnedByUs = false;
notifyFillBufferDone(outHeader);
outHeader = NULL;
outInfo = NULL;
mInputSize = 0;
}
}
int main(int argc, char **argv)
{
wav_io_context_t wav_io = { read_callback, seek_callback, tell_callback };
m4af_io_callbacks_t
m4af_io = { read_callback, write_callback, seek_callback, tell_callback };
aacenc_param_ex_t params = { 0 };
int result = 2;
FILE *ifp = 0;
FILE *ofp = 0;
char *output_filename = 0;
#ifdef USE_LIBSNDFILE
SNDFILE* snd = NULL;
SF_INFO snd_info;
pcm_sample_description_t snd_desc = { 0 };
#else
wav_reader_t *wavf = 0;
#endif
HANDLE_AACENCODER encoder = 0;
AACENC_InfoStruct aacinfo = { 0 };
m4af_ctx_t *m4af = 0;
const pcm_sample_description_t *sample_format;
int downsampled_timescale = 0;
int frame_count = 0;
struct stat stb = { 0 };
setlocale(LC_CTYPE, "");
setbuf(stderr, 0);
if (parse_options(argc, argv, ¶ms) < 0)
return 1;
#ifdef USE_LIBSNDFILE
if ((snd = sf_open (params.input_filename, SFM_READ, &snd_info)) == NULL) {
fprintf(stderr, "ERROR: broken / unsupported input file\n");
goto END;
}
#ifdef USE_LIBSAMPLERATE
if(params.resample) {
snd_desc.sample_rate = params.resample;
printf("resampling to %dhz\n", snd_desc.sample_rate);
} else {
snd_desc.sample_rate = snd_info.samplerate;
}
snd_desc.sample_type = PCM_TYPE_FLOAT; // always -- libsndfile does the conversion for us
snd_desc.bits_per_channel = sizeof(float)*8;
#else
snd_desc.sample_rate = snd_info.samplerate;
snd_desc.sample_type = PCM_TYPE_SINT; // always -- libsndfile does the conversion for us
snd_desc.bits_per_channel = sizeof(short)*8;
#endif
snd_desc.channels_per_frame = snd_info.channels;
snd_desc.bytes_per_frame = snd_info.channels * (snd_desc.bits_per_channel / 8);
snd_desc.channel_mask = 0;
sample_format = &snd_desc;
#else
if ((ifp = aacenc_fopen(params.input_filename, "rb")) == 0) {
aacenc_fprintf(stderr, "ERROR: %s: %s\n", params.input_filename,
strerror(errno));
goto END;
}
if (fstat(fileno(ifp), &stb) == 0 && (stb.st_mode & S_IFMT) != S_IFREG) {
wav_io.seek = 0;
wav_io.tell = 0;
}
if (!params.is_raw) {
if ((wavf = wav_open(&wav_io, ifp, params.ignore_length)) == 0) {
fprintf(stderr, "ERROR: broken / unsupported input file\n");
goto END;
}
} else {
int bytes_per_channel;
pcm_sample_description_t desc = { 0 };
if (parse_raw_spec(params.raw_format, &desc) < 0) {
fprintf(stderr, "ERROR: invalid raw-format spec\n");
goto END;
}
desc.sample_rate = params.raw_rate;
desc.channels_per_frame = params.raw_channels;
bytes_per_channel = (desc.bits_per_channel + 7) / 8;
desc.bytes_per_frame = params.raw_channels * bytes_per_channel;
if ((wavf = raw_open(&wav_io, ifp, &desc)) == 0) {
fprintf(stderr, "ERROR: failed to open raw input\n");
goto END;
}
}
sample_format = wav_get_format(wavf);
#endif
if (aacenc_init(&encoder, (aacenc_param_t*)¶ms, sample_format,
&aacinfo) < 0)
goto END;
if (!params.output_filename) {
const char *ext = params.transport_format ? ".aac" : ".m4a";
output_filename = generate_output_filename(params.input_filename, ext);
params.output_filename = output_filename;
}
if ((ofp = aacenc_fopen(params.output_filename, "wb+")) == 0) {
aacenc_fprintf(stderr, "ERROR: %s: %s\n", params.output_filename,
strerror(errno));
goto END;
}
handle_signals();
if (!params.transport_format) {
uint32_t scale;
unsigned framelen = aacinfo.frameLength;
int sbr_mode = aacenc_is_sbr_active((aacenc_param_t*)¶ms);
int sig_mode = aacEncoder_GetParam(encoder, AACENC_SIGNALING_MODE);
if (sbr_mode && !sig_mode)
downsampled_timescale = 1;
scale = sample_format->sample_rate >> downsampled_timescale;
if ((m4af = m4af_create(M4AF_CODEC_MP4A, scale, &m4af_io, ofp)) < 0)
goto END;
m4af_set_decoder_specific_info(m4af, 0, aacinfo.confBuf,
aacinfo.confSize);
m4af_set_fixed_frame_duration(m4af, 0,
framelen >> downsampled_timescale);
m4af_set_priming_mode(m4af, params.gapless_mode + 1);
m4af_begin_write(m4af);
}
