mirror of
https://github.com/mpv-player/mpv
synced 2024-12-26 09:02:38 +00:00
1ed6e96cfb
common functions for channel reordering. This fixes these modules by adding channel reordering code for 5.0/5.1 audio: ao: pcm ad: dmo, faad, ffmpeg(ac3, dca, libfaad, liba52), pcm ae: faac, lavc(ac3, libfaac), pcm git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@25343 b3059339-0415-0410-9bf9-f77b7e298cf2
84 lines
2.3 KiB
C
84 lines
2.3 KiB
C
#include <stdio.h>
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#include <stdlib.h>
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#include <inttypes.h>
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#include <unistd.h>
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#include <string.h>
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#include <sys/types.h>
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#include "m_option.h"
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#include "mp_msg.h"
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#include "libmpdemux/aviheader.h"
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#include "libaf/af_format.h"
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#include "libaf/reorder_ch.h"
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#include "libmpdemux/ms_hdr.h"
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#include "stream/stream.h"
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#include "libmpdemux/muxer.h"
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#include "ae_pcm.h"
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static int bind_pcm(audio_encoder_t *encoder, muxer_stream_t *mux_a)
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{
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mux_a->h.dwScale=1;
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mux_a->h.dwRate=encoder->params.sample_rate;
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mux_a->wf=malloc(sizeof(WAVEFORMATEX));
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mux_a->wf->wFormatTag=0x1; // PCM
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mux_a->wf->nChannels=encoder->params.channels;
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mux_a->h.dwSampleSize=2*mux_a->wf->nChannels;
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mux_a->wf->nBlockAlign=mux_a->h.dwSampleSize;
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mux_a->wf->nSamplesPerSec=mux_a->h.dwRate;
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mux_a->wf->nAvgBytesPerSec=mux_a->h.dwSampleSize*mux_a->wf->nSamplesPerSec;
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mux_a->wf->wBitsPerSample=16;
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mux_a->wf->cbSize=0; // FIXME for l3codeca.acm
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encoder->input_format = (mux_a->wf->wBitsPerSample==8) ? AF_FORMAT_U8 : AF_FORMAT_S16_LE;
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encoder->min_buffer_size = 16384;
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encoder->max_buffer_size = mux_a->wf->nAvgBytesPerSec;
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return 1;
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}
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static int encode_pcm(audio_encoder_t *encoder, uint8_t *dest, void *src, int nsamples, int max_size)
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{
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max_size = FFMIN(nsamples, max_size);
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if (encoder->params.channels == 6 || encoder->params.channels == 5) {
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max_size -= max_size % (encoder->params.channels * 2);
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reorder_channel_copy_nch(src, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
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dest, AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
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encoder->params.channels,
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max_size / 2, 2);
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}
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else
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memcpy(dest, src, max_size);
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return max_size;
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}
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static int set_decoded_len(audio_encoder_t *encoder, int len)
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{
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return len;
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}
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static int close_pcm(audio_encoder_t *encoder)
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{
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return 1;
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}
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static int get_frame_size(audio_encoder_t *encoder)
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{
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return 0;
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}
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int mpae_init_pcm(audio_encoder_t *encoder)
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{
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encoder->params.samples_per_frame = encoder->params.sample_rate;
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encoder->params.bitrate = encoder->params.sample_rate * encoder->params.channels * 2 * 8;
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encoder->decode_buffer_size = encoder->params.bitrate / 8;
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encoder->bind = bind_pcm;
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encoder->get_frame_size = get_frame_size;
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encoder->set_decoded_len = set_decoded_len;
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encoder->encode = encode_pcm;
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encoder->close = close_pcm;
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return 1;
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}
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