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| author | wm4 <wm4@nowhere> | 2015-03-10 15:12:52 +0100 |
|---|---|---|
| committer | wm4 <wm4@nowhere> | 2015-03-10 15:12:52 +0100 |
| commit | 5f8b060ec2dde7d0ff5468a5b7499851c22b756d (patch) | |
| tree | 593321dd494d3bf24727ec7ee3a7631b07b49cc8 | |
| parent | 574bd127ecadb6d7cbcad9aea2187b05a433dda1 (diff) | |
| download | mpv-5f8b060ec2dde7d0ff5468a5b7499851c22b756d.tar.bz2 mpv-5f8b060ec2dde7d0ff5468a5b7499851c22b756d.tar.xz | |
ad_spdif: move frame sizes to a general function
Needed for the next commit. This commit should probably be reverted as
soon as we're working with full audio frames internally, instead of
"flat" FIFOs.
| -rw-r--r-- | audio/decode/ad_spdif.c | 8 | ||||
| -rw-r--r-- | audio/format.c | 13 | ||||
| -rw-r--r-- | audio/format.h | 2 |
3 files changed, 16 insertions, 7 deletions
diff --git a/audio/decode/ad_spdif.c b/audio/decode/ad_spdif.c index f7891767d2..3da5def868 100644 --- a/audio/decode/ad_spdif.c +++ b/audio/decode/ad_spdif.c @@ -117,13 +117,11 @@ static int init(struct dec_audio *da, const char *decoder) int samplerate = 0; switch (stream->codec->codec_id) { case AV_CODEC_ID_AAC: - spdif_ctx->iec61937_packet_size = 16384; sample_format = AF_FORMAT_S_AAC; samplerate = 48000; num_channels = 2; break; case AV_CODEC_ID_AC3: - spdif_ctx->iec61937_packet_size = 6144; sample_format = AF_FORMAT_S_AC3; samplerate = 48000; num_channels = 2; @@ -131,31 +129,26 @@ static int init(struct dec_audio *da, const char *decoder) case AV_CODEC_ID_DTS: if (da->opts->dtshd) { av_dict_set(&format_opts, "dtshd_rate", "768000", 0); // 4*192000 - spdif_ctx->iec61937_packet_size = 32768; sample_format = AF_FORMAT_S_DTSHD; samplerate = 192000; num_channels = 2*4; } else { - spdif_ctx->iec61937_packet_size = 32768; sample_format = AF_FORMAT_S_DTS; samplerate = 48000; num_channels = 2; } break; case AV_CODEC_ID_EAC3: - spdif_ctx->iec61937_packet_size = 24576; sample_format = AF_FORMAT_S_EAC3; samplerate = 192000; num_channels = 2; break; case AV_CODEC_ID_MP3: - spdif_ctx->iec61937_packet_size = 4608; sample_format = AF_FORMAT_S_MP3; samplerate = 48000; num_channels = 2; break; case AV_CODEC_ID_TRUEHD: - spdif_ctx->iec61937_packet_size = 61440; sample_format = AF_FORMAT_S_TRUEHD; samplerate = 192000; num_channels = 8; @@ -166,6 +159,7 @@ static int init(struct dec_audio *da, const char *decoder) mp_audio_set_num_channels(&spdif_ctx->fmt, num_channels); mp_audio_set_format(&spdif_ctx->fmt, sample_format); spdif_ctx->fmt.rate = samplerate; + spdif_ctx->iec61937_packet_size = af_format_sample_alignment(sample_format); if (avformat_write_header(lavf_ctx, &format_opts) < 0) { MP_FATAL(da, "libavformat spdif initialization failed.\n"); diff --git a/audio/format.c b/audio/format.c index 08b369c090..5e83ead1b2 100644 --- a/audio/format.c +++ b/audio/format.c @@ -227,3 +227,16 @@ int af_format_conversion_score(int dst_format, int src_format) score -= 2048; // has to convert float<->int return score; } + +int af_format_sample_alignment(int format) +{ + switch (format) { + case AF_FORMAT_S_AAC: return 16384; + case AF_FORMAT_S_AC3: return 6144; + case AF_FORMAT_S_DTSHD: return 32768; + case AF_FORMAT_S_EAC3: return 24576; + case AF_FORMAT_S_MP3: return 4608; + case AF_FORMAT_S_TRUEHD: return 61440; + default: return 1; + } +} diff --git a/audio/format.h b/audio/format.h index b3ac0000c8..910b952fd2 100644 --- a/audio/format.h +++ b/audio/format.h @@ -125,4 +125,6 @@ void af_fill_silence(void *dst, size_t bytes, int format); int af_format_conversion_score(int dst_format, int src_format); +int af_format_sample_alignment(int format); + #endif /* MPLAYER_AF_FORMAT_H */ |