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audio: improve decoder open failure handling

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Reinitialize sh_audio->samplesize and sample_format before falling back
to another audio decoder (some decoders rely on default values). Remove
code setting these fields from demux_mkv and demux_lavf (no decoder
should depend on demuxer-set values for these fields).

Conflicts:
	audio/decode/ad_lavc.c

Merged from mplayer2 commit 6b9567. The changes to ad_lavc.c are not
merged, as they are very specific to the mplayer2 libavresample hack;
we deplanarize manually, so we can't get unsupported sample formats
yet (except on raw audio with "pcm_f64le", as we don't support
AV_SAMPLE_FMT_DBL in the audio chain).
This commit is contained in:
Uoti Urpala 2012-11-26 04:47:29 +02:00 committed by wm4
parent a75e65db5d
commit 77eac2ec34
3 changed files with 4 additions and 15 deletions
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2
audio/decode/dec_audio.c
View File

@ -64,6 +64,8 @@ static int init_audio_codec(sh_audio_t *sh_audio)
{
assert(!sh_audio->initialized);
resync_audio_stream(sh_audio);
sh_audio->samplesize = 2;
sh_audio->sample_format = AF_FORMAT_S16_NE;
if ((af_cfg.force & AF_INIT_FORMAT_MASK) == AF_INIT_FLOAT) {
int fmt = AF_FORMAT_FLOAT_NE;
if (sh_audio->ad_driver->control(sh_audio, ADCTRL_QUERY_FORMAT,

10
demux/demux_lavf.c
View File

@ -389,16 +389,6 @@ static void handle_stream(demuxer_t *demuxer, AVFormatContext *avfc, int i)
sh_audio->samplerate = codec->sample_rate;
sh_audio->i_bps = codec->bit_rate / 8;
switch (codec->codec_id) {
case CODEC_ID_PCM_S8:
case CODEC_ID_PCM_U8:
sh_audio->samplesize = 1;
break;
case CODEC_ID_PCM_S16LE:
case CODEC_ID_PCM_S16BE:
case CODEC_ID_PCM_U16LE:
case CODEC_ID_PCM_U16BE:
sh_audio->samplesize = 2;
break;
case CODEC_ID_PCM_ALAW:
sh_audio->format = 0x6;
break;

7
demux/demux_mkv.c
View File

@ -1365,13 +1365,10 @@ static int demux_mkv_open_audio(demuxer_t *demuxer, mkv_track_t *track,
sh_a->samplerate = (uint32_t) track->a_sfreq;
sh_a->container_out_samplerate = track->a_osfreq;
sh_a->wf->nSamplesPerSec = (uint32_t) track->a_sfreq;
if (track->a_bps == 0) {
sh_a->samplesize = 2;
if (track->a_bps == 0)
sh_a->wf->wBitsPerSample = 16;
} else {
sh_a->samplesize = track->a_bps / 8;
else
sh_a->wf->wBitsPerSample = track->a_bps;
}
if (track->a_formattag == 0x0055) { /* MP3 || MP2 */
sh_a->wf->nAvgBytesPerSec = 16000;
sh_a->wf->nBlockAlign = 1152;