audio: support parameter changes (e.g. channel count) during playback

Add support for parameter changes (e.g. channel count) during playback.
This makes decoding AC3 files that switch between 2 and 6 channels
work reasonably well even with -channels 6 and ffac3 decoder.

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@31737 b3059339-0415-0410-9bf9-f77b7e298cf2

Fix typo in error message: ACC -> AAC

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32473 b3059339-0415-0410-9bf9-f77b7e298cf2

Avoid printing AAC with SBR warning on every decode call, instead print
it only after every decoder reconfiguration.

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32476 b3059339-0415-0410-9bf9-f77b7e298cf2

2 files changed, 61 insertions, 25 deletions

diff --git a/libmpcodecs/ad_ffmpeg.c b/libmpcodecs/ad_ffmpeg.c
index 8578314bc6..6d27a314f6 100644
--- a/libmpcodecs/ad_ffmpeg.c
+++ b/libmpcodecs/ad_ffmpeg.c
@@ -52,6 +52,44 @@ static int preinit(sh_audio_t *sh)
   return 1;
 }
 
+static int setup_format(sh_audio_t *sh_audio, const AVCodecContext *lavc_context)
+{
+    int broken_srate = 0;
+    int samplerate    = lavc_context->sample_rate;
+    int sample_format = sh_audio->sample_format;
+    switch (lavc_context->sample_fmt) {
+        case SAMPLE_FMT_U8:  sample_format = AF_FORMAT_U8;       break;
+        case SAMPLE_FMT_S16: sample_format = AF_FORMAT_S16_NE;   break;
+        case SAMPLE_FMT_S32: sample_format = AF_FORMAT_S32_NE;   break;
+        case SAMPLE_FMT_FLT: sample_format = AF_FORMAT_FLOAT_NE; break;
+        default:
+            mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Unsupported sample format\n");
+    }
+    if(sh_audio->wf){
+        // If the decoder uses the wrong number of channels all is lost anyway.
+        // sh_audio->channels=sh_audio->wf->nChannels;
+
+        if (lavc_context->codec_id == CODEC_ID_AAC &&
+            samplerate == 2*sh_audio->wf->nSamplesPerSec) {
+            broken_srate = 1;
+        } else if (sh_audio->wf->nSamplesPerSec)
+            samplerate=sh_audio->wf->nSamplesPerSec;
+    }
+    if (lavc_context->channels != sh_audio->channels ||
+        samplerate != sh_audio->samplerate ||
+        sample_format != sh_audio->sample_format) {
+        sh_audio->channels=lavc_context->channels;
+        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;
+}
+
 static int init(sh_audio_t *sh_audio)
 {
     struct MPOpts *opts = sh_audio->opts;
@@ -133,32 +171,19 @@ static int init(sh_audio_t *sh_audio)
    } while (x <= 0 && tries++ < 5);
    if(x>0) sh_audio->a_buffer_len=x;
 
-  sh_audio->channels=lavc_context->channels;
-  sh_audio->samplerate=lavc_context->sample_rate;
   sh_audio->i_bps=lavc_context->bit_rate/8;
+  if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec)
+      sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
+
   switch (lavc_context->sample_fmt) {
-      case SAMPLE_FMT_U8:  sh_audio->sample_format = AF_FORMAT_U8;       break;
-      case SAMPLE_FMT_S16: sh_audio->sample_format = AF_FORMAT_S16_NE;   break;
-      case SAMPLE_FMT_S32: sh_audio->sample_format = AF_FORMAT_S32_NE;   break;
-      case SAMPLE_FMT_FLT: sh_audio->sample_format = AF_FORMAT_FLOAT_NE; break;
+      case SAMPLE_FMT_U8:
+      case SAMPLE_FMT_S16:
+      case SAMPLE_FMT_S32:
+      case SAMPLE_FMT_FLT:
+          break;
       default:
-          mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Unsupported sample format\n");
           return 0;
   }
-  /* If the audio is AAC the container level data may be unreliable
-   * because of SBR handling problems (possibly half real sample rate at
-   * container level). Default AAC decoding with ad_faad has used codec-level
-   * values for a long time without generating complaints so it should be OK.
-   */
-  if (sh_audio->wf && lavc_context->codec_id != CODEC_ID_AAC) {
-      // If the decoder uses the wrong number of channels all is lost anyway.
-      // sh_audio->channels=sh_audio->wf->nChannels;
-      if (sh_audio->wf->nSamplesPerSec)
-      sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
-      if (sh_audio->wf->nAvgBytesPerSec)
-      sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
-  }
-  sh_audio->samplesize=af_fmt2bits(sh_audio->sample_format)/ 8;
   return 1;
 }
 
@@ -232,6 +257,9 @@ static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int m
 	  sh_audio->pts_bytes += len2;
 	}
         mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d  \n",y,len2);
+
+        if (setup_format(sh_audio, sh_audio->context))
+            break;
     }
   return len;
 }
diff --git a/libmpcodecs/dec_audio.c b/libmpcodecs/dec_audio.c
index 218203272d..ade2429dc0 100644
--- a/libmpcodecs/dec_audio.c
+++ b/libmpcodecs/dec_audio.c
@@ -378,13 +378,20 @@ static int filter_n_bytes(sh_audio_t *sh, int len)
     int error = 0;
 
     // Decode more bytes if needed
+    int old_samplerate = sh->samplerate;
+    int 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;
 	int minlen = len - sh->a_buffer_len;
 	int maxlen = sh->a_buffer_size - sh->a_buffer_len;
 	int ret = sh->ad_driver->decode_audio(sh, buf, minlen, maxlen);
-	if (ret <= 0) {
-	    error = -1;
+	int format_change = sh->samplerate != old_samplerate
+                            || sh->channels != old_channels
+                            || sh->sample_format != old_sample_format;
+	if (ret <= 0 || format_change) {
+	    error = format_change ? -2 : -1;
+            // samples from format-changing call get discarded too
 	    len = sh->a_buffer_len;
 	    break;
 	}
@@ -467,8 +474,9 @@ int decode_audio(sh_audio_t *sh_audio, int minlen)
 	    /* if this iteration does not fill buffer, we must have lots
 	     * of buffering in filters */
 	    huge_filter_buffer = 1;
-	if (filter_n_bytes(sh_audio, declen) < 0)
-	    return -1;
+	int res = filter_n_bytes(sh_audio, declen);
+	if (res < 0)
+	    return res;
     }
     return 0;
 }