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| author | Warren Dukes <warren.dukes@gmail.com> | 2004-02-26 03:21:24 +0000 |
|---|---|---|
| committer | Warren Dukes <warren.dukes@gmail.com> | 2004-02-26 03:21:24 +0000 |
| commit | 1adca2abd2aff8e74691d180f00600c3467ded28 (patch) | |
| tree | 7a4d22a7ff842b1b17466c8d6f31c5aa13b2c293 /src | |
| parent | 9d4bc4efea0359896863b67e5e5223d8025099f8 (diff) | |
| download | mpd-1adca2abd2aff8e74691d180f00600c3467ded28.tar.gz mpd-1adca2abd2aff8e74691d180f00600c3467ded28.tar.xz mpd-1adca2abd2aff8e74691d180f00600c3467ded28.zip | |
add bitrate reporting to flac
git-svn-id: https://svn.musicpd.org/mpd/trunk@63 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Diffstat (limited to 'src')
| -rw-r--r-- | src/flac_decode.c | 19 |
1 files changed, 17 insertions, 2 deletions
diff --git a/src/flac_decode.c b/src/flac_decode.c index 2708795cf..fd4969073 100644 --- a/src/flac_decode.c +++ b/src/flac_decode.c @@ -33,6 +33,8 @@ typedef struct { unsigned char chunk[CHUNK_SIZE]; int chunk_length; float time; + int bitRate; + FLAC__uint64 position; Buffer * cb; AudioFormat * af; DecoderControl * dc; @@ -57,6 +59,8 @@ void flacPlayFile(char *file, Buffer * cb, AudioFormat * af, data.chunk_length = 0; data.time = 0; + data.position = 0; + data.bitRate = 0; data.cb = cb; data.af = af; data.dc = dc; @@ -106,6 +110,7 @@ void flacPlayFile(char *file, Buffer * cb, AudioFormat * af, { data.time = ((float)sampleToSeek)/ af->sampleRate; + data.position = 0; } dc->seek = 0; } @@ -189,7 +194,7 @@ int flacSendChunk(FlacData * data) { CHUNK_SIZE); data->cb->chunkSize[data->cb->end] = data->chunk_length; data->cb->times[data->cb->end] = data->time; - data->cb->bitRate[data->cb->end] = 0; + data->cb->bitRate[data->cb->end] = data->bitRate; data->cb->end++; if(data->cb->end>=buffered_chunks) { @@ -207,8 +212,18 @@ FLAC__StreamDecoderWriteStatus flacWrite(const FLAC__FileDecoder *dec, const FLA unsigned char * uc; int c_samp, c_chan, d_samp; int i; + float timeChange; + FLAC__uint64 newPosition = 0; - data->time+=((float)samples)/frame->header.sample_rate; + timeChange = ((float)samples)/frame->header.sample_rate; + data->time+= timeChange; + + FLAC__file_decoder_get_decode_position(dec,&newPosition); + if(data->position) { + data->bitRate = ((newPosition-data->position)*8.0/timeChange) + /1024+0.5; + } + data->position = newPosition; for(c_samp = d_samp = 0; c_samp < frame->header.blocksize; c_samp++) { for(c_chan = 0; c_chan < frame->header.channels; |