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fix indentation

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This commit is contained in:
mchinen
2012-06-01 06:11:26 +00:00
parent 80a2b68bcf
commit f837c1fefd
1 changed files with 32 additions and 32 deletions
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64
src/ondemand/ODDecodeFFmpegTask.cpp
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@@ -289,7 +289,7 @@ int ODFFmpegDecoder::Decode(samplePtr & data, sampleFormat & format, sampleCount
int stindex = -1;
uint64_t targetts;
//printf("attempting seek to %llu\n", start);
printf("attempting seek to %llu\n", start);
//we have to find the index for this stream.
for (unsigned int i = 0; i < mFormatContext->nb_streams; i++) {
if (mFormatContext->streams[i] == sc->m_stream )
@@ -332,44 +332,44 @@ int ODFFmpegDecoder::Decode(samplePtr & data, sampleFormat & format, sampleCount
if (sc != (streamContext*)1)
{
nChannels = sc->m_stream->codec->channels < sc->m_initialchannels ? sc->m_stream->codec->channels : sc->m_initialchannels;
//find out the dts we've seekd to. can't use the stream->cur_dts because it is faulty. also note that until we do the first seek, pkt.dts can be false and will change for the same samples after the initial seek.
sampleCount actualDecodeStart = 0.52 + sc->m_stream->codec->sample_rate * sc->m_pkt.dts * ((double)sc->m_stream->time_base.num/sc->m_stream->time_base.den); //this is mostly safe because den is usually 1 or low number but check for high values.
//find out the dts we've seekd to. can't use the stream->cur_dts because it is faulty. also note that until we do the first seek, pkt.dts can be false and will change for the same samples after the initial seek.
sampleCount actualDecodeStart = 0.52 + sc->m_stream->codec->sample_rate * sc->m_pkt.dts * ((double)sc->m_stream->time_base.num/sc->m_stream->time_base.den); //this is mostly safe because den is usually 1 or low number but check for high values.
//hack to get rounding to work to neareset frame size since dts isn't exact
if (sc->m_stream->codec->frame_size) {
actualDecodeStart = ((actualDecodeStart + sc->m_stream->codec->frame_size/2) / sc->m_stream->codec->frame_size) * sc->m_stream->codec->frame_size;
}
//hack to get rounding to work to neareset frame size since dts isn't exact
if (sc->m_stream->codec->frame_size) {
actualDecodeStart = ((actualDecodeStart + sc->m_stream->codec->frame_size/2) / sc->m_stream->codec->frame_size) * sc->m_stream->codec->frame_size;
}
if(actualDecodeStart != mCurrentPos)
//printf("ts not matching - now:%llu (%f), last:%llu, lastlen:%llu, start %llu, len %llu\n",actualDecodeStart, actualDecodeStartdouble, mCurrentPos, mCurrentLen, start, len);
//if we've skipped over some samples, fill the gap with silence. This could happen often in the beginning of the file.
if(actualDecodeStart>start && firstpass) {
// find the number of samples for the leading silence
int amt = actualDecodeStart - start;
FFMpegDecodeCache* cache = new FFMpegDecodeCache;
if(actualDecodeStart != mCurrentPos)
printf("ts not matching - now:%llu , last:%llu, lastlen:%llu, start %llu, len %llu\n",actualDecodeStart, mCurrentPos, mCurrentLen, start, len);
//if we've skipped over some samples, fill the gap with silence. This could happen often in the beginning of the file.
if(actualDecodeStart>start && firstpass) {
// find the number of samples for the leading silence
int amt = actualDecodeStart - start;
FFMpegDecodeCache* cache = new FFMpegDecodeCache;
//printf("skipping/zeroing %i samples. - now:%llu (%f), last:%llu, lastlen:%llu, start %llu, len %llu\n",amt,actualDecodeStart, actualDecodeStartdouble, mCurrentPos, mCurrentLen, start, len);
//printf("skipping/zeroing %i samples. - now:%llu (%f), last:%llu, lastlen:%llu, start %llu, len %llu\n",amt,actualDecodeStart, actualDecodeStartdouble, mCurrentPos, mCurrentLen, start, len);
//put it in the cache so the other channels can use it.
cache->numChannels = sc->m_stream->codec->channels;
cache->len = amt;
cache->start=start;
// 8 bit and 16 bit audio output from ffmpeg means
// 16 bit int out.
// 32 bit int, float, double mean float out.
if (format == int16Sample)
cache->samplefmt = SAMPLE_FMT_S16;
else
cache->samplefmt = SAMPLE_FMT_FLT;
//put it in the cache so the other channels can use it.
cache->numChannels = sc->m_stream->codec->channels;
cache->len = amt;
cache->start=start;
// 8 bit and 16 bit audio output from ffmpeg means
// 16 bit int out.
// 32 bit int, float, double mean float out.
if (format == int16Sample)
cache->samplefmt = SAMPLE_FMT_S16;
else
cache->samplefmt = SAMPLE_FMT_FLT;
cache->samplePtr = (uint8_t*) malloc(amt * cache->numChannels * SAMPLE_SIZE(format));
cache->samplePtr = (uint8_t*) malloc(amt * cache->numChannels * SAMPLE_SIZE(format));
memset(cache->samplePtr, 0, amt * cache->numChannels * SAMPLE_SIZE(format));
memset(cache->samplePtr, 0, amt * cache->numChannels * SAMPLE_SIZE(format));
InsertCache(cache);
}
firstpass=false;
mCurrentPos = actualDecodeStart;
InsertCache(cache);
}
firstpass=false;
mCurrentPos = actualDecodeStart;
//decode the entire packet (unused bits get saved in cache, so as long as cache size limit is bigger than the
//largest packet size, we're ok.
while (sc->m_pktRemainingSiz > 0)