diff options
Diffstat (limited to 'libmpcodecs/ad_sample.c')
| -rw-r--r-- | libmpcodecs/ad_sample.c | 24 |
1 files changed, 12 insertions, 12 deletions
diff --git a/libmpcodecs/ad_sample.c b/libmpcodecs/ad_sample.c index 848152c68e..5ddf4ef77b 100644 --- a/libmpcodecs/ad_sample.c +++ b/libmpcodecs/ad_sample.c @@ -23,18 +23,18 @@ static int preinit(sh_audio_t *sh){ // let's check if the driver is available, return 0 if not. // (you should do that if you use external lib(s) which is optional) ... - + // there are default values set for buffering, but you can override them: - + // minimum output buffer size (should be the uncompressed max. frame size) sh->audio_out_minsize=4*2*1024; // in this sample, we assume max 4 channels, // 2 bytes/sample and 1024 samples/frame // Default: 8192 - + // minimum input buffer size (set only if you need input buffering) // (should be the max compressed frame size) sh->audio_in_minsize=2048; // Default: 0 (no input buffer) - + // if you set audio_in_minsize non-zero, the buffer will be allocated // before the init() call by the core, and you can access it via // pointer: sh->audio_in_buffer @@ -43,17 +43,17 @@ static int preinit(sh_audio_t *sh){ // the next few parameters define the audio format (channels, sample type, // in/out bitrate etc.). it's OK to move these to init() if you can set // them only after some initialization: - + sh->samplesize=2; // bytes (not bits!) per sample per channel sh->channels=2; // number of channels sh->samplerate=44100; // samplerate sh->sample_format=AF_FORMAT_S16_LE; // sample format, see libao2/afmt.h - + sh->i_bps=64000/8; // input data rate (compressed bytes per second) // Note: if you have VBR or unknown input rate, set it to some common or // average value, instead of zero. it's used to predict time delay of // buffered compressed bytes, so it must be more-or-less real! - + //sh->o_bps=... // output data rate (uncompressed bytes per second) // Note: you DON'T need to set o_bps in most cases, as it defaults to: // sh->samplesize*sh->channels*sh->samplerate; @@ -62,7 +62,7 @@ static int preinit(sh_audio_t *sh){ // set the compressed and uncompressed packet size (used by the demuxer): sh->ds->ss_mul = 34; // compressed packet size sh->ds->ss_div = 64; // samples per packet - + return 1; // return values: 1=OK 0=ERROR } @@ -71,7 +71,7 @@ static int init(sh_audio_t *sh_audio){ // you can store HANDLE or private struct pointer at sh->context // you can access WAVEFORMATEX header at sh->wf - + // set sample format/rate parameters if you didn't do it in preinit() yet. return 1; // return values: 1=OK 0=ERROR @@ -86,7 +86,7 @@ static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int m // audio decoding. the most important thing :) // parameters you get: - // buf = pointer to the output buffer, you have to store uncompressed + // buf = pointer to the output buffer, you have to store uncompressed // samples there // minlen = requested minimum size (in bytes!) of output. it's just a // _recommendation_, you can decode more or less, it just tell you that @@ -96,8 +96,8 @@ static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int m // buffer, it's the upper-most limit! // note: maxlen will be always greater or equal to sh->audio_out_minsize - // now, let's decode... - + // now, let's decode... + // you can read the compressed stream using the demux stream functions: // demux_read_data(sh->ds, buffer, length) - read 'length' bytes to 'buffer' // ds_get_packet(sh->ds, &buffer) - set ptr buffer to next data packet |