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author | wm4 <wm4@nowhere> | 2013-04-06 22:43:12 +0200 |
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committer | wm4 <wm4@nowhere> | 2013-05-12 21:24:55 +0200 |
commit | 4b5cee4617d0decbf93d06df4f45097fd7e00105 (patch) | |
tree | b6d293e86d83ab5c3efd99153aa7a7a134d2937f /audio/decode/ad_spdif.c | |
parent | 586b75ad0840e154835ae67c7720b71bd36f8cc9 (diff) | |
download | mpv-4b5cee4617d0decbf93d06df4f45097fd7e00105.tar.bz2 mpv-4b5cee4617d0decbf93d06df4f45097fd7e00105.tar.xz |
core: use channel map on demuxer level too
This helps passing the channel layout correctly from decoder to audio
filter chain. (Because that part "reuses" the demuxer level codec
parameters, which is very disgusting.)
Note that ffmpeg stuff already passed the channel layout via
mp_copy_lav_codec_headers(). So other than easier dealing with the
demuxer/decoder parameters mess, there's no real advantage to doing
this.
Make the --channels option accept a channel map. Since simple numbers
map to standard layouts with the given number of channels, this is
downwards compatible. Likewise for demux_rawaudio.
Diffstat (limited to 'audio/decode/ad_spdif.c')
-rw-r--r-- | audio/decode/ad_spdif.c | 17 |
1 files changed, 10 insertions, 7 deletions
diff --git a/audio/decode/ad_spdif.c b/audio/decode/ad_spdif.c index ad735dde7d..a6f41932e9 100644 --- a/audio/decode/ad_spdif.c +++ b/audio/decode/ad_spdif.c @@ -148,19 +148,20 @@ static int init(sh_audio_t *sh, const char *decoder) } sh->ds->buffer_pos -= in_size; + int num_channels = 0; switch (lavf_ctx->streams[0]->codec->codec_id) { case AV_CODEC_ID_AAC: spdif_ctx->iec61937_packet_size = 16384; sh->sample_format = AF_FORMAT_IEC61937_LE; sh->samplerate = srate; - sh->channels = 2; + num_channels = 2; sh->i_bps = bps; break; case AV_CODEC_ID_AC3: spdif_ctx->iec61937_packet_size = 6144; sh->sample_format = AF_FORMAT_AC3_LE; sh->samplerate = srate; - sh->channels = 2; + num_channels = 2; sh->i_bps = bps; break; case AV_CODEC_ID_DTS: @@ -175,13 +176,13 @@ static int init(sh_audio_t *sh, const char *decoder) spdif_ctx->iec61937_packet_size = 32768; sh->sample_format = AF_FORMAT_IEC61937_LE; sh->samplerate = 192000; // DTS core require 48000 - sh->channels = 2*4; + num_channels = 2*4; sh->i_bps = bps; } else { spdif_ctx->iec61937_packet_size = 32768; sh->sample_format = AF_FORMAT_AC3_LE; sh->samplerate = srate; - sh->channels = 2; + num_channels = 2; sh->i_bps = bps; } break; @@ -189,26 +190,28 @@ static int init(sh_audio_t *sh, const char *decoder) spdif_ctx->iec61937_packet_size = 24576; sh->sample_format = AF_FORMAT_IEC61937_LE; sh->samplerate = 192000; - sh->channels = 2; + num_channels = 2; sh->i_bps = bps; break; case AV_CODEC_ID_MP3: spdif_ctx->iec61937_packet_size = 4608; sh->sample_format = AF_FORMAT_MPEG2; sh->samplerate = srate; - sh->channels = 2; + num_channels = 2; sh->i_bps = bps; break; case AV_CODEC_ID_TRUEHD: spdif_ctx->iec61937_packet_size = 61440; sh->sample_format = AF_FORMAT_IEC61937_LE; sh->samplerate = 192000; - sh->channels = 8; + num_channels = 8; sh->i_bps = bps; break; default: break; } + if (num_channels) + mp_chmap_from_channels(&sh->channels, num_channels); return 1; |