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-rw-r--r--audio/decode/ad_lavc.c18
-rw-r--r--audio/decode/ad_mpg123.c2
-rw-r--r--audio/decode/ad_spdif.c17
-rw-r--r--audio/decode/dec_audio.c18
4 files changed, 34 insertions, 21 deletions
diff --git a/audio/decode/ad_lavc.c b/audio/decode/ad_lavc.c
index 8177d9cde6..4997a66bc4 100644
--- a/audio/decode/ad_lavc.c
+++ b/audio/decode/ad_lavc.c
@@ -168,10 +168,16 @@ static int setup_format(sh_audio_t *sh_audio,
else if (container_samplerate)
samplerate = container_samplerate;
- if (lavc_context->channels != sh_audio->channels ||
+ struct mp_chmap lavc_chmap;
+ mp_chmap_from_lavc(&lavc_chmap, lavc_context->channel_layout);
+ // No channel layout or layout disagrees with channel count
+ if (lavc_chmap.num != lavc_context->channels)
+ mp_chmap_from_channels(&lavc_chmap, lavc_context->channels);
+
+ if (!mp_chmap_equals(&lavc_chmap, &sh_audio->channels) ||
samplerate != sh_audio->samplerate ||
sample_format != sh_audio->sample_format) {
- sh_audio->channels = lavc_context->channels;
+ sh_audio->channels = lavc_chmap;
sh_audio->samplerate = samplerate;
sh_audio->sample_format = sample_format;
sh_audio->samplesize = af_fmt2bits(sh_audio->sample_format) / 8;
@@ -227,8 +233,11 @@ static int init(sh_audio_t *sh_audio, const char *decoder)
lavc_context->codec_type = AVMEDIA_TYPE_AUDIO;
lavc_context->codec_id = lavc_codec->id;
- if (opts->downmix)
- lavc_context->request_channels = mpopts->audio_output_channels;
+ if (opts->downmix) {
+ lavc_context->request_channels = mpopts->audio_output_channels.num;
+ lavc_context->request_channel_layout =
+ mp_chmap_to_lavc(&mpopts->audio_output_channels);
+ }
// Always try to set - option only exists for AC3 at the moment
av_opt_set_double(lavc_context, "drc_scale", opts->ac3drc,
@@ -246,6 +255,7 @@ static int init(sh_audio_t *sh_audio, const char *decoder)
lavc_context->codec_tag = sh_audio->format;
lavc_context->sample_rate = sh_audio->samplerate;
lavc_context->bit_rate = sh_audio->i_bps * 8;
+ lavc_context->channel_layout = mp_chmap_to_lavc(&sh_audio->channels);
if (sh_audio->wf)
set_from_wf(lavc_context, sh_audio->wf);
diff --git a/audio/decode/ad_mpg123.c b/audio/decode/ad_mpg123.c
index 999dc2fbba..45538f42f6 100644
--- a/audio/decode/ad_mpg123.c
+++ b/audio/decode/ad_mpg123.c
@@ -358,7 +358,7 @@ static int init(sh_audio_t *sh, const char *decoder)
con->mean_count = 0;
#endif
con->vbr = (finfo.vbr != MPG123_CBR);
- sh->channels = channels;
+ mp_chmap_from_channels(&sh->channels, channels);
sh->samplerate = rate;
/* Without external force, mpg123 will always choose signed encoding,
* and non-16-bit only on builds that don't support it.
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;
diff --git a/audio/decode/dec_audio.c b/audio/decode/dec_audio.c
index 11232f9271..999a96a10b 100644
--- a/audio/decode/dec_audio.c
+++ b/audio/decode/dec_audio.c
@@ -86,7 +86,7 @@ static int init_audio_codec(sh_audio_t *sh_audio, const char *decoder)
sh_audio->initialized = 1;
- if (!sh_audio->channels || !sh_audio->samplerate) {
+ if (mp_chmap_is_empty(&sh_audio->channels) || !sh_audio->samplerate) {
mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Audio decoder did not specify "
"audio format!\n");
uninit_audio(sh_audio); // free buffers
@@ -94,7 +94,7 @@ static int init_audio_codec(sh_audio_t *sh_audio, const char *decoder)
}
if (!sh_audio->o_bps)
- sh_audio->o_bps = sh_audio->channels * sh_audio->samplerate
+ sh_audio->o_bps = sh_audio->channels.num * sh_audio->samplerate
* sh_audio->samplesize;
return 1;
}
@@ -160,14 +160,14 @@ int init_best_audio_codec(sh_audio_t *sh_audio, char *audio_decoders)
sh_audio->gsh->decoder_desc);
mp_msg(MSGT_DECAUDIO, MSGL_V,
"AUDIO: %d Hz, %d ch, %s, %3.1f kbit/%3.2f%% (ratio: %d->%d)\n",
- sh_audio->samplerate, sh_audio->channels,
+ sh_audio->samplerate, sh_audio->channels.num,
af_fmt2str_short(sh_audio->sample_format),
sh_audio->i_bps * 8 * 0.001,
((float) sh_audio->i_bps / sh_audio->o_bps) * 100.0,
sh_audio->i_bps, sh_audio->o_bps);
mp_msg(MSGT_IDENTIFY, MSGL_INFO,
"ID_AUDIO_BITRATE=%d\nID_AUDIO_RATE=%d\n" "ID_AUDIO_NCH=%d\n",
- sh_audio->i_bps * 8, sh_audio->samplerate, sh_audio->channels);
+ sh_audio->i_bps * 8, sh_audio->samplerate, sh_audio->channels.num);
} else {
mp_msg(MSGT_DECAUDIO, MSGL_ERR,
"Failed to initialize an audio decoder for codec '%s'.\n",
@@ -207,7 +207,7 @@ int init_audio_filters(sh_audio_t *sh_audio, int in_samplerate,
afs = af_new(sh_audio->opts);
// input format: same as codec's output format:
afs->input.rate = in_samplerate;
- mp_audio_set_num_channels(&afs->input, sh_audio->channels);
+ mp_audio_set_channels(&afs->input, &sh_audio->channels);
mp_audio_set_format(&afs->input, sh_audio->sample_format);
// output format: same as ao driver's input format (if missing, fallback to input)
@@ -259,7 +259,7 @@ static int filter_n_bytes(sh_audio_t *sh, struct bstr *outbuf, int len)
// Decode more bytes if needed
int old_samplerate = sh->samplerate;
- int old_channels = sh->channels;
+ struct mp_chmap old_channels = sh->channels;
int old_sample_format = sh->sample_format;
while (sh->a_buffer_len < len) {
unsigned char *buf = sh->a_buffer + sh->a_buffer_len;
@@ -267,7 +267,7 @@ static int filter_n_bytes(sh_audio_t *sh, struct bstr *outbuf, int len)
int maxlen = sh->a_buffer_size - sh->a_buffer_len;
int ret = sh->ad_driver->decode_audio(sh, buf, minlen, maxlen);
int format_change = sh->samplerate != old_samplerate
- || sh->channels != old_channels
+ || !mp_chmap_equals(&sh->channels, &old_channels)
|| sh->sample_format != old_sample_format;
if (ret <= 0 || format_change) {
error = format_change ? -2 : -1;
@@ -285,7 +285,7 @@ static int filter_n_bytes(sh_audio_t *sh, struct bstr *outbuf, int len)
.rate = sh->samplerate,
};
mp_audio_set_format(&filter_input, sh->sample_format);
- mp_audio_set_num_channels(&filter_input, sh->channels);
+ mp_audio_set_channels(&filter_input, &sh->channels);
struct mp_audio *filter_output = af_play(sh->afilter, &filter_input);
if (!filter_output)
@@ -314,7 +314,7 @@ int decode_audio(sh_audio_t *sh_audio, struct bstr *outbuf, int minlen)
// Indicates that a filter seems to be buffering large amounts of data
int huge_filter_buffer = 0;
// Decoded audio must be cut at boundaries of this many bytes
- int unitsize = sh_audio->channels * sh_audio->samplesize * 16;
+ int unitsize = sh_audio->channels.num * sh_audio->samplesize * 16;
/* Filter output size will be about filter_multiplier times input size.
* If some filter buffers audio in big blocks this might only hold
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-18 22:22:56 +0000