audio: don't write decoded audio format to sh_audio

sh_audio is supposed to contain file headers, not whatever was decoded.
Fix this, and write the decoded format to separate fields in the decoder
context, the dec_audio.decoded field. (Note that this field is really
only needed to communicate the audio format from decoder driver to the
generic code, so no other code accesses it.)

5 files changed, 42 insertions, 47 deletions

diff --git a/audio/decode/ad_lavc.c b/audio/decode/ad_lavc.c
index 143990414b..8663d5fc32 100644
--- a/audio/decode/ad_lavc.c
+++ b/audio/decode/ad_lavc.c
@@ -161,9 +161,9 @@ static int setup_format(struct dec_audio *da)
             lavc_chmap = sh_audio->channels;
     }
 
-    sh_audio->channels = lavc_chmap;
-    sh_audio->samplerate = samplerate;
-    sh_audio->sample_format = sample_format;
+    mp_audio_set_channels(&da->decoded, &lavc_chmap);
+    mp_audio_set_format(&da->decoded, sample_format);
+    da->decoded.rate = samplerate;
     return 0;
 }
 
@@ -366,9 +366,7 @@ static int decode_new_packet(struct dec_audio *da)
         return -1;
 
     priv->frame.samples = priv->avframe->nb_samples;
-    mp_audio_set_format(&priv->frame, da->header->audio->sample_format);
-    mp_audio_set_channels(&priv->frame, &da->header->audio->channels);
-    priv->frame.rate = da->header->audio->samplerate;
+    mp_audio_copy_config(&priv->frame, &da->decoded);
     for (int n = 0; n < priv->frame.num_planes; n++)
         priv->frame.planes[n] = priv->avframe->data[n];
 
diff --git a/audio/decode/ad_mpg123.c b/audio/decode/ad_mpg123.c
index 1abda4e084..8f168ebb95 100644
--- a/audio/decode/ad_mpg123.c
+++ b/audio/decode/ad_mpg123.c
@@ -151,8 +151,8 @@ static int set_format(struct dec_audio *da)
     int encoding;
     ret = mpg123_getformat(con->handle, &rate, &channels, &encoding);
     if (ret == MPG123_OK) {
-        mp_chmap_from_channels(&da->header->audio->channels, channels);
-        da->header->audio->samplerate = rate;
+        mp_audio_set_num_channels(&da->decoded, channels);
+        da->decoded.rate = rate;
         int af = mpg123_format_to_af(encoding);
         if (!af) {
             /* This means we got a funny custom build of libmpg123 that only supports an unknown format. */
@@ -160,7 +160,7 @@ static int set_format(struct dec_audio *da)
                    "Bad encoding from mpg123: %i.\n", encoding);
             return MPG123_ERR;
         }
-        da->header->audio->sample_format = af;
+        mp_audio_set_format(&da->decoded, af);
         con->sample_size = channels * (af_fmt2bits(af) / 8);
         con->new_format = 0;
     }
@@ -308,9 +308,7 @@ static int decode_audio(struct dec_audio *da, struct mp_audio *buffer, int maxle
             return -1;
     }
 
-    if (da->header->audio->samplerate != buffer->rate ||
-        !mp_chmap_equals(&da->header->audio->channels, &buffer->channels) ||
-        da->header->audio->sample_format != buffer->format)
+    if (!mp_audio_config_equals(&da->decoded, buffer))
         return 0;
 
     size_t got_now = 0;
diff --git a/audio/decode/ad_spdif.c b/audio/decode/ad_spdif.c
index 2630cc89b9..e0c0e88671 100644
--- a/audio/decode/ad_spdif.c
+++ b/audio/decode/ad_spdif.c
@@ -116,57 +116,59 @@ static int init(struct dec_audio *da, const char *decoder)
     AVDictionary *format_opts = NULL;
 
     int num_channels = 0;
-    struct sh_audio *sh = da->header->audio;
+    int sample_format = 0;
+    int samplerate = 0;
     switch (stream->codec->codec_id) {
     case AV_CODEC_ID_AAC:
         spdif_ctx->iec61937_packet_size = 16384;
-        sh->sample_format               = AF_FORMAT_IEC61937_LE;
-        sh->samplerate                  = 48000;
+        sample_format                   = AF_FORMAT_IEC61937_LE;
+        samplerate                      = 48000;
         num_channels                    = 2;
         break;
     case AV_CODEC_ID_AC3:
         spdif_ctx->iec61937_packet_size = 6144;
-        sh->sample_format               = AF_FORMAT_AC3_LE;
-        sh->samplerate                  = 48000;
+        sample_format                   = AF_FORMAT_AC3_LE;
+        samplerate                      = 48000;
         num_channels                    = 2;
         break;
     case AV_CODEC_ID_DTS:
-        if (sh->opts->dtshd) {
+        if (da->opts->dtshd) {
             av_dict_set(&format_opts, "dtshd_rate", "768000", 0); // 4*192000
             spdif_ctx->iec61937_packet_size = 32768;
-            sh->sample_format               = AF_FORMAT_IEC61937_LE;
-            sh->samplerate                  = 192000; // DTS core require 48000
+            sample_format                   = AF_FORMAT_IEC61937_LE;
+            samplerate                      = 192000;
             num_channels                    = 2*4;
         } else {
             spdif_ctx->iec61937_packet_size = 32768;
-            sh->sample_format               = AF_FORMAT_AC3_LE;
-            sh->samplerate                  = 48000;
+            sample_format                   = AF_FORMAT_AC3_LE;
+            samplerate                      = 48000;
             num_channels                    = 2;
         }
         break;
     case AV_CODEC_ID_EAC3:
         spdif_ctx->iec61937_packet_size = 24576;
-        sh->sample_format               = AF_FORMAT_IEC61937_LE;
-        sh->samplerate                  = 192000;
+        sample_format                   = AF_FORMAT_IEC61937_LE;
+        samplerate                      = 192000;
         num_channels                    = 2;
         break;
     case AV_CODEC_ID_MP3:
         spdif_ctx->iec61937_packet_size = 4608;
-        sh->sample_format               = AF_FORMAT_MPEG2;
-        sh->samplerate                  = 48000;
+        sample_format                   = AF_FORMAT_MPEG2;
+        samplerate                      = 48000;
         num_channels                    = 2;
         break;
     case AV_CODEC_ID_TRUEHD:
         spdif_ctx->iec61937_packet_size = 61440;
-        sh->sample_format               = AF_FORMAT_IEC61937_LE;
-        sh->samplerate                  = 192000;
+        sample_format                   = AF_FORMAT_IEC61937_LE;
+        samplerate                      = 192000;
         num_channels                    = 8;
         break;
     default:
         abort();
     }
-    if (num_channels)
-        mp_chmap_from_channels(&sh->channels, num_channels);
+    mp_audio_set_num_channels(&da->decoded, num_channels);
+    mp_audio_set_format(&da->decoded, sample_format);
+    da->decoded.rate = samplerate;
 
     if (avformat_write_header(lavf_ctx, &format_opts) < 0) {
         mp_msg(MSGT_DECAUDIO, MSGL_FATAL,
@@ -190,7 +192,7 @@ static int decode_audio(struct dec_audio *da, struct mp_audio *buffer, int maxle
     struct spdifContext *spdif_ctx = da->priv;
     AVFormatContext     *lavf_ctx  = spdif_ctx->lavf_ctx;
 
-    int sstride = 2 * da->header->audio->channels.num;
+    int sstride = 2 * da->decoded.channels.num;
     assert(sstride == buffer->sstride);
 
     if (maxlen * sstride < spdif_ctx->iec61937_packet_size)
diff --git a/audio/decode/dec_audio.c b/audio/decode/dec_audio.c
index 39ee3d5695..566bade875 100644
--- a/audio/decode/dec_audio.c
+++ b/audio/decode/dec_audio.c
@@ -66,9 +66,7 @@ static const struct ad_functions * const ad_drivers[] = {
 // Drop audio buffer and reinit it (after format change)
 static void reinit_audio_buffer(struct dec_audio *da)
 {
-    struct sh_audio *sh = da->header->audio;
-    mp_audio_buffer_reinit_fmt(da->decode_buffer, sh->sample_format,
-                               &sh->channels, sh->samplerate);
+    mp_audio_buffer_reinit(da->decode_buffer, &da->decoded);
     mp_audio_buffer_preallocate_min(da->decode_buffer, DECODE_BUFFER_SAMPLES);
 }
 
@@ -100,9 +98,8 @@ static int init_audio_codec(struct dec_audio *d_audio, const char *decoder)
 
     d_audio->initialized = 1;
 
-    struct sh_audio *sh = d_audio->header->audio;
-    if (mp_chmap_is_empty(&sh->channels) || !sh->samplerate ||
-        !sh->sample_format)
+    if (!d_audio->decoded.channels.num || !d_audio->decoded.rate ||
+        !d_audio->decoded.format)
     {
         mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Audio decoder did not specify "
                 "audio format!\n");
@@ -177,12 +174,12 @@ int audio_init_best_codec(struct dec_audio *d_audio, char *audio_decoders)
                d_audio->decoder_desc);
         mp_msg(MSGT_DECAUDIO, MSGL_V,
                "AUDIO: %d Hz, %d ch, %s\n",
-               d_audio->header->audio->samplerate, d_audio->header->audio->channels.num,
-               af_fmt_to_str(d_audio->header->audio->sample_format));
+               d_audio->decoded.rate, d_audio->decoded.channels.num,
+               af_fmt_to_str(d_audio->decoded.format));
         mp_msg(MSGT_IDENTIFY, MSGL_INFO,
                "ID_AUDIO_BITRATE=%d\nID_AUDIO_RATE=%d\n" "ID_AUDIO_NCH=%d\n",
-               d_audio->i_bps * 8, d_audio->header->audio->samplerate,
-               d_audio->header->audio->channels.num);
+               d_audio->i_bps * 8, d_audio->decoded.rate,
+               d_audio->decoded.channels.num);
     } else {
         mp_msg(MSGT_DECAUDIO, MSGL_ERR,
                "Failed to initialize an audio decoder for codec '%s'.\n",
@@ -269,11 +266,7 @@ static int filter_n_bytes(struct dec_audio *da, struct mp_audio_buffer *outbuf,
         // Commit the data just read as valid data
         mp_audio_buffer_finish_write(da->decode_buffer, buffer.samples);
         // Format change
-        struct sh_audio *sh = da->header->audio;
-        if (sh->samplerate != config.rate ||
-            !mp_chmap_equals(&sh->channels, &config.channels) ||
-            sh->sample_format != config.format)
-        {
+        if (!mp_audio_config_equals(&da->decoded, &config)) {
             // If there are still samples left in the buffer, let them drain
             // first, and don't signal a format change to the caller yet.
             if (mp_audio_buffer_samples(da->decode_buffer) > 0)
diff --git a/audio/decode/dec_audio.h b/audio/decode/dec_audio.h
index bc5c3793d8..be5fc44dfc 100644
--- a/audio/decode/dec_audio.h
+++ b/audio/decode/dec_audio.h
@@ -20,6 +20,7 @@
 #define MPLAYER_DEC_AUDIO_H
 
 #include "audio/chmap.h"
+#include "audio/audio.h"
 #include "demux/stheader.h"
 
 struct mp_audio_buffer;
@@ -34,7 +35,10 @@ struct dec_audio {
     int initialized;
     char *decoder_desc;
     // set by decoder
-    int i_bps;          // input bitrate
+    struct mp_audio decoded;    // format of decoded audio (no data, temporarily
+                                // different from decode_buffer during format
+                                // changes)
+    int i_bps;                  // input bitrate, can change with VBR sources
     // last known pts value in output from decoder
     double pts;
     // number of samples output by decoder after last known pts