mirror of
https://github.com/mpv-player/mpv
synced 2025-03-04 13:18:12 +00:00
ad_pcm used the old audio timestamp tracking system that calculated timestamp at end of decoder output as last_timestamp_in_input_decoder_has_read + bytes_read_after_that_timestamp / input_bitrate. For PCM this can be accurate as input bitrate is constant. However it relies on input bitrate being known and actually set. At least in some case with .mov input and libavformat demuxer it wasn't set. Instead of special-casing PCM to make sure input bitrate is set (in general it may not be known or constant at all) change ad_pcm to explicitly set the pts information on the decoder output side.
171 lines
5.2 KiB
C
171 lines
5.2 KiB
C
#include <stdio.h>
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#include <stdlib.h>
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#include <unistd.h>
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#include "talloc.h"
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#include "config.h"
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#include "ad_internal.h"
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#include "libaf/af_format.h"
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#include "libaf/reorder_ch.h"
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static const ad_info_t info =
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{
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"Uncompressed PCM audio decoder",
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"pcm",
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"Nick Kurshev",
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"A'rpi",
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""
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};
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struct ad_pcm_context {
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unsigned char *packet_ptr;
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int packet_len;
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};
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LIBAD_EXTERN(pcm)
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static int init(sh_audio_t *sh_audio)
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{
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WAVEFORMATEX *h=sh_audio->wf;
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sh_audio->i_bps=h->nAvgBytesPerSec;
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sh_audio->channels=h->nChannels;
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sh_audio->samplerate=h->nSamplesPerSec;
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sh_audio->samplesize=(h->wBitsPerSample+7)/8;
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sh_audio->sample_format=AF_FORMAT_S16_LE; // default
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switch(sh_audio->format){ /* hardware formats: */
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case 0x0:
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case 0x1: // Microsoft PCM
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case 0xfffe: // Extended
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switch (sh_audio->samplesize) {
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case 1: sh_audio->sample_format=AF_FORMAT_U8; break;
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case 2: sh_audio->sample_format=AF_FORMAT_S16_LE; break;
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case 3: sh_audio->sample_format=AF_FORMAT_S24_LE; break;
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case 4: sh_audio->sample_format=AF_FORMAT_S32_LE; break;
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}
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break;
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case 0x3: // IEEE float
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sh_audio->sample_format=AF_FORMAT_FLOAT_LE;
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break;
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case 0x6: sh_audio->sample_format=AF_FORMAT_A_LAW;break;
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case 0x7: sh_audio->sample_format=AF_FORMAT_MU_LAW;break;
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case 0x11: sh_audio->sample_format=AF_FORMAT_IMA_ADPCM;break;
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case 0x50: sh_audio->sample_format=AF_FORMAT_MPEG2;break;
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/* case 0x2000: sh_audio->sample_format=AFMT_AC3; */
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case 0x20776172: // 'raw '
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sh_audio->sample_format=AF_FORMAT_S16_BE;
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if(sh_audio->samplesize==1) sh_audio->sample_format=AF_FORMAT_U8;
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break;
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case 0x736F7774: // 'twos'
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sh_audio->sample_format=AF_FORMAT_S16_BE;
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// intended fall-through
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case 0x74776F73: // 'sowt'
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if(sh_audio->samplesize==1) sh_audio->sample_format=AF_FORMAT_S8;
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break;
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case 0x32336c66: // 'fl32', bigendian float32
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sh_audio->sample_format=AF_FORMAT_FLOAT_BE;
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sh_audio->samplesize=4;
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break;
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case 0x666c3332: // '23lf', little endian float32, MPlayer internal fourCC
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sh_audio->sample_format=AF_FORMAT_FLOAT_LE;
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sh_audio->samplesize=4;
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break;
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/* case 0x34366c66: // 'fl64', bigendian float64
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sh_audio->sample_format=AF_FORMAT_FLOAT_BE;
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sh_audio->samplesize=8;
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break;
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case 0x666c3634: // '46lf', little endian float64, MPlayer internal fourCC
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sh_audio->sample_format=AF_FORMAT_FLOAT_LE;
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sh_audio->samplesize=8;
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break;*/
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case 0x34326e69: // 'in24', bigendian int24
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sh_audio->sample_format=AF_FORMAT_S24_BE;
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sh_audio->samplesize=3;
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break;
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case 0x696e3234: // '42ni', little endian int24, MPlayer internal fourCC
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sh_audio->sample_format=AF_FORMAT_S24_LE;
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sh_audio->samplesize=3;
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break;
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case 0x32336e69: // 'in32', bigendian int32
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sh_audio->sample_format=AF_FORMAT_S32_BE;
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sh_audio->samplesize=4;
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break;
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case 0x696e3332: // '23ni', little endian int32, MPlayer internal fourCC
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sh_audio->sample_format=AF_FORMAT_S32_LE;
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sh_audio->samplesize=4;
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break;
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default: if(sh_audio->samplesize!=2) sh_audio->sample_format=AF_FORMAT_U8;
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}
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if (!sh_audio->samplesize) // this would cause MPlayer to hang later
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sh_audio->samplesize = 2;
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sh_audio->context = talloc_zero(NULL, struct ad_pcm_context);
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return 1;
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}
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static int preinit(sh_audio_t *sh)
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{
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sh->audio_out_minsize=2048;
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return 1;
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}
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static void uninit(sh_audio_t *sh)
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{
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talloc_free(sh->context);
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}
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static int control(sh_audio_t *sh,int cmd,void* arg, ...)
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{
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int skip;
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switch(cmd)
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{
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case ADCTRL_SKIP_FRAME:
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skip=sh->i_bps/16;
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skip=skip&(~3);
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demux_read_data(sh->ds,NULL,skip);
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return CONTROL_TRUE;
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}
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return CONTROL_UNKNOWN;
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}
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static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
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{
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unsigned len = sh_audio->channels*sh_audio->samplesize;
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minlen = (minlen + len - 1) / len * len;
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if (minlen > maxlen)
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// if someone needs hundreds of channels adjust audio_out_minsize
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// based on channels in preinit()
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return -1;
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len = 0;
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struct ad_pcm_context *ctx = sh_audio->context;
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while (len < minlen) {
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if (ctx->packet_len == 0) {
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double pts;
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int plen = ds_get_packet_pts(sh_audio->ds, &ctx->packet_ptr, &pts);
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if (plen < 0)
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break;
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ctx->packet_len = plen;
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if (pts != MP_NOPTS_VALUE) {
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sh_audio->pts = pts;
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sh_audio->pts_bytes = 0;
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}
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}
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int from_stored = ctx->packet_len;
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if (from_stored > minlen - len)
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from_stored = minlen - len;
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memcpy(buf + len, ctx->packet_ptr, from_stored);
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ctx->packet_len -= from_stored;
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ctx->packet_ptr += from_stored;
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sh_audio->pts_bytes += from_stored;
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len += from_stored;
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}
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if (len == 0)
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len = -1; // The loop above only exits at error/EOF
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if (len > 0 && sh_audio->channels >= 5) {
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reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
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AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
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sh_audio->channels,
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len / sh_audio->samplesize, sh_audio->samplesize);
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}
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return len;
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}
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