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.

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