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mirror of https://github.com/cookiengineer/audacity synced 2025-08-09 16:41:14 +02:00

fix indentation

This commit is contained in:
mchinen 2012-06-01 06:11:26 +00:00
parent 80a2b68bcf
commit f837c1fefd

View File

@ -289,7 +289,7 @@ int ODFFmpegDecoder::Decode(samplePtr & data, sampleFormat & format, sampleCount
int stindex = -1; int stindex = -1;
uint64_t targetts; 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. //we have to find the index for this stream.
for (unsigned int i = 0; i < mFormatContext->nb_streams; i++) { for (unsigned int i = 0; i < mFormatContext->nb_streams; i++) {
if (mFormatContext->streams[i] == sc->m_stream ) if (mFormatContext->streams[i] == sc->m_stream )
@ -332,44 +332,44 @@ int ODFFmpegDecoder::Decode(samplePtr & data, sampleFormat & format, sampleCount
if (sc != (streamContext*)1) if (sc != (streamContext*)1)
{ {
nChannels = sc->m_stream->codec->channels < sc->m_initialchannels ? sc->m_stream->codec->channels : sc->m_initialchannels; 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. //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. 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 //hack to get rounding to work to neareset frame size since dts isn't exact
if (sc->m_stream->codec->frame_size) { 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; actualDecodeStart = ((actualDecodeStart + sc->m_stream->codec->frame_size/2) / sc->m_stream->codec->frame_size) * sc->m_stream->codec->frame_size;
} }
if(actualDecodeStart != mCurrentPos) 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); 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 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) { if(actualDecodeStart>start && firstpass) {
// find the number of samples for the leading silence // find the number of samples for the leading silence
int amt = actualDecodeStart - start; int amt = actualDecodeStart - start;
FFMpegDecodeCache* cache = new FFMpegDecodeCache; 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. //put it in the cache so the other channels can use it.
cache->numChannels = sc->m_stream->codec->channels; cache->numChannels = sc->m_stream->codec->channels;
cache->len = amt; cache->len = amt;
cache->start=start; cache->start=start;
// 8 bit and 16 bit audio output from ffmpeg means // 8 bit and 16 bit audio output from ffmpeg means
// 16 bit int out. // 16 bit int out.
// 32 bit int, float, double mean float out. // 32 bit int, float, double mean float out.
if (format == int16Sample) if (format == int16Sample)
cache->samplefmt = SAMPLE_FMT_S16; cache->samplefmt = SAMPLE_FMT_S16;
else else
cache->samplefmt = SAMPLE_FMT_FLT; 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); InsertCache(cache);
} }
firstpass=false; firstpass=false;
mCurrentPos = actualDecodeStart; mCurrentPos = actualDecodeStart;
//decode the entire packet (unused bits get saved in cache, so as long as cache size limit is bigger than the //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. //largest packet size, we're ok.
while (sc->m_pktRemainingSiz > 0) while (sc->m_pktRemainingSiz > 0)