demux_mkv: never force output sample rate

Matroska has an output sample rate (OutputSamplingFrequency), which in
theory should be forced instead of whatever the decoder outputs. But it
appears no software (other than mplayer2 and mpv until now) actually
respects this. Even worse, there were broken files around, which played
correctly with (in theory) broken software, but not mplayer2/mpv. Hacks
were added to our code to play these files correctly, but they didn't
catch all cases.

Simplify this by doing what everyone else does, and always use the
decoder's sample rate instead. In particular, we try to handle all
sample rate issues like libavformat's Matroska demuxer does.

audio/decode/ad_lavc.c

@@ -160,16 +160,12 @@ static int setup_format(sh_audio_t *sh_audio,
     struct priv *priv = sh_audio->context;
     int sample_format        =
         af_from_avformat(av_get_packed_sample_fmt(lavc_context->sample_fmt));
-    bool broken_srate        = false;
     int samplerate           = lavc_context->sample_rate;
-    int container_samplerate = sh_audio->container_out_samplerate;
+    // If not set, try container samplerate
-    if (!container_samplerate && sh_audio->wf)
+    if (!samplerate && sh_audio->wf) {
-        container_samplerate = sh_audio->wf->nSamplesPerSec;
+        samplerate = sh_audio->wf->nSamplesPerSec;
-    if (lavc_context->codec_id == AV_CODEC_ID_AAC
+        mp_tmsg(MSGT_DECAUDIO, MSGL_V, "ad_lavc: using container rate.\n");
-        && samplerate == 2 * container_samplerate)
+    }
-        broken_srate = true;
-    else if (container_samplerate)
-        samplerate = container_samplerate;
 
     struct mp_chmap lavc_chmap;
     mp_chmap_from_lavc(&lavc_chmap, lavc_context->channel_layout);
@@ -188,9 +184,6 @@ static int setup_format(sh_audio_t *sh_audio,
         sh_audio->samplerate = samplerate;
         sh_audio->sample_format = sample_format;
         sh_audio->samplesize = af_fmt2bits(sh_audio->sample_format) / 8;
-        if (broken_srate)
-            mp_msg(MSGT_DECAUDIO, MSGL_WARN,
-                   "Ignoring broken container sample rate for AAC with SBR\n");
         return 1;
     }
     return 0;

demux/demux_mkv.c

@@ -488,14 +488,6 @@ static void parse_trackaudio(struct demuxer *demuxer, struct mkv_track *track,
                "[mkv] |   + Output sampling frequency: %f\n", track->a_osfreq);
     } else
         track->a_osfreq = track->a_sfreq;
-    // Something creates files with osfreq incorrectly set
-    if (track->a_sfreq == 44100 && track->a_osfreq == 96000) {
-        mp_msg(MSGT_DEMUX, MSGL_WARN, "[mkv] Audio track has codec frequency "
-               "%.1f and playback frequency %.1f.\n[mkv] This looks wrong. "
-               "Assuming this file is corrupt and ignoring the latter.\n",
-               track->a_sfreq, track->a_osfreq);
-        track->a_osfreq = track->a_sfreq;
-    }
     if (audio->n_bit_depth) {
         track->a_bps = audio->bit_depth;
         mp_msg(MSGT_DEMUX, MSGL_V, "[mkv] |   + Bit depth: %u\n",
@@ -1359,9 +1351,12 @@ static int demux_mkv_open_audio(demuxer_t *demuxer, mkv_track_t *track)
     sh_a->gsh->demuxer_id = track->tnum;
     sh_a->gsh->title = talloc_strdup(sh_a, track->name);
     sh_a->gsh->default_track = track->default_track;
+    if (!track->a_osfreq)
+        track->a_osfreq = track->a_sfreq;
     if (track->ms_compat) {
         if (track->private_size < sizeof(*sh_a->wf))
             goto error;
+        mp_msg(MSGT_DEMUX, MSGL_V, "[mkv] track with MS compat audio.\n");
         WAVEFORMATEX *wf = (WAVEFORMATEX *) track->private_data;
         sh_a->wf = calloc(1, track->private_size);
         sh_a->wf->wFormatTag = le2me_16(wf->wFormatTag);
@@ -1373,8 +1368,8 @@ static int demux_mkv_open_audio(demuxer_t *demuxer, mkv_track_t *track)
         sh_a->wf->cbSize = track->private_size - sizeof(*sh_a->wf);
         memcpy(sh_a->wf + 1, wf + 1,
                track->private_size - sizeof(*sh_a->wf));
-        if (track->a_sfreq == 0.0)
+        if (track->a_osfreq == 0.0)
-            track->a_sfreq = sh_a->wf->nSamplesPerSec;
+            track->a_osfreq = sh_a->wf->nSamplesPerSec;
         if (track->a_channels == 0)
             track->a_channels = sh_a->wf->nChannels;
         if (track->a_bps == 0)
@@ -1402,9 +1397,8 @@ static int demux_mkv_open_audio(demuxer_t *demuxer, mkv_track_t *track)
     sh_a->wf->wFormatTag = track->a_formattag;
     mp_chmap_from_channels(&sh_a->channels, track->a_channels);
     sh_a->wf->nChannels = track->a_channels;
-    sh_a->samplerate = (uint32_t) track->a_sfreq;
+    sh_a->samplerate = (uint32_t) track->a_osfreq;
-    sh_a->container_out_samplerate = track->a_osfreq;
+    sh_a->wf->nSamplesPerSec = (uint32_t) track->a_osfreq;
-    sh_a->wf->nSamplesPerSec = (uint32_t) track->a_sfreq;
     if (track->a_bps == 0)
         sh_a->wf->wBitsPerSample = 16;
     else
@@ -1438,7 +1432,7 @@ static int demux_mkv_open_audio(demuxer_t *demuxer, mkv_track_t *track)
         } else {
             /* Recreate the 'private data' */
             /* which faad2 uses in its initialization */
-            srate_idx = aac_get_sample_rate_index(sh_a->samplerate);
+            srate_idx = aac_get_sample_rate_index(track->a_sfreq);
             if (!strncmp(&track->codec_id[12], "MAIN", 4))
                 profile = 0;
             else if (!strncmp(&track->codec_id[12], "LC", 2))
@@ -1456,8 +1450,7 @@ static int demux_mkv_open_audio(demuxer_t *demuxer, mkv_track_t *track)
                 /* HE-AAC (aka SBR AAC) */
                 sh_a->codecdata_len = 5;
 
-                sh_a->samplerate *= 2;
+                sh_a->samplerate = sh_a->wf->nSamplesPerSec = track->a_osfreq;
-                sh_a->wf->nSamplesPerSec *= 2;
                 srate_idx = aac_get_sample_rate_index(sh_a->samplerate);
                 sh_a->codecdata[2] = AAC_SYNC_EXTENSION_TYPE >> 3;
                 sh_a->codecdata[3] = ((AAC_SYNC_EXTENSION_TYPE & 0x07) << 5) | 5;
@@ -1587,7 +1580,7 @@ static int demux_mkv_open_audio(demuxer_t *demuxer, mkv_track_t *track)
         AV_WL16(data + 4, 1);
         AV_WL16(data + 6, sh_a->channels.num);
         AV_WL16(data + 8, sh_a->wf->wBitsPerSample);
-        AV_WL32(data + 10, sh_a->samplerate);
+        AV_WL32(data + 10, track->a_osfreq);
         // Bogus: last frame won't be played.
         AV_WL32(data + 14, 0);
     } else if (!track->ms_compat) {

demux/stheader.h

@@ -90,7 +90,6 @@ typedef struct sh_audio {
     // output format:
     int sample_format;
     int samplerate;
-    int container_out_samplerate;
     int samplesize;
     struct mp_chmap channels;
     int i_bps; // == bitrate  (compressed bytes/sec)