Mercurial > mplayer.hg
view libmpcodecs/ae_pcm.c @ 24892:80180dc13565
Change decode_audio() interface
Rewrite decode_audio to better deal with filters that handle input in
large blocks. It now always places output in sh_audio->a_out_buffer
(which was always given as a parameter before) and reallocates the
buffer if needed. After the changes filters can return arbitrarily
large blocks of data without some of it being lost. The new version
also allows simplifying some code.
author | uau |
---|---|
date | Thu, 01 Nov 2007 06:52:19 +0000 |
parents | ed8f90096c65 |
children | dfa8a510c81c |
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#include <stdio.h> #include <stdlib.h> #include <inttypes.h> #include <unistd.h> #include <string.h> #include <sys/types.h> #include "m_option.h" #include "mp_msg.h" #include "libmpdemux/aviheader.h" #include "libaf/af_format.h" #include "libmpdemux/ms_hdr.h" #include "stream/stream.h" #include "libmpdemux/muxer.h" #include "ae_pcm.h" static int bind_pcm(audio_encoder_t *encoder, muxer_stream_t *mux_a) { mux_a->h.dwScale=1; mux_a->h.dwRate=encoder->params.sample_rate; mux_a->wf=malloc(sizeof(WAVEFORMATEX)); mux_a->wf->wFormatTag=0x1; // PCM mux_a->wf->nChannels=encoder->params.channels; mux_a->h.dwSampleSize=2*mux_a->wf->nChannels; mux_a->wf->nBlockAlign=mux_a->h.dwSampleSize; mux_a->wf->nSamplesPerSec=mux_a->h.dwRate; mux_a->wf->nAvgBytesPerSec=mux_a->h.dwSampleSize*mux_a->wf->nSamplesPerSec; mux_a->wf->wBitsPerSample=16; mux_a->wf->cbSize=0; // FIXME for l3codeca.acm encoder->input_format = (mux_a->wf->wBitsPerSample==8) ? AF_FORMAT_U8 : AF_FORMAT_S16_LE; encoder->min_buffer_size = 16384; encoder->max_buffer_size = mux_a->wf->nAvgBytesPerSec; return 1; } static int encode_pcm(audio_encoder_t *encoder, uint8_t *dest, void *src, int nsamples, int max_size) { max_size = FFMIN(nsamples, max_size); memcpy(dest, src, max_size); return max_size; } static int set_decoded_len(audio_encoder_t *encoder, int len) { return len; } static int close_pcm(audio_encoder_t *encoder) { return 1; } static int get_frame_size(audio_encoder_t *encoder) { return 0; } int mpae_init_pcm(audio_encoder_t *encoder) { encoder->params.samples_per_frame = encoder->params.sample_rate; encoder->params.bitrate = encoder->params.sample_rate * encoder->params.channels * 2 * 8; encoder->decode_buffer_size = encoder->params.bitrate / 8; encoder->bind = bind_pcm; encoder->get_frame_size = get_frame_size; encoder->set_decoded_len = set_decoded_len; encoder->encode = encode_pcm; encoder->close = close_pcm; return 1; }