[SCM] libav/experimental: Fix encoding when the input audio format/rate/channels changes during transcoding.

[siretart at users.alioth.debian.org]

The following commit has been merged in the experimental branch:
commit 8afab6860512eaaaad37d89b796eaee6c4659771
Author: Stefano Sabatini <stefano.sabatini-lala at poste.it>
Date:   Sun Dec 12 16:18:50 2010 +0000

    Fix encoding when the input audio format/rate/channels changes during
    transcoding.
    
    Fix issue #2292.
    
    Patch sponsored by KIM Keep In Mind GmbH, srl.
    
    Originally committed as revision 25939 to svn://svn.ffmpeg.org/ffmpeg/trunk

diff --git a/ffmpeg.c b/ffmpeg.c
index 143ec8f..4107eed 100644
--- a/ffmpeg.c
+++ b/ffmpeg.c
@@ -295,6 +295,9 @@ typedef struct AVOutputStream {
     /* audio only */
     int audio_resample;
     ReSampleContext *resample; /* for audio resampling */
+    int resample_sample_fmt;
+    int resample_channels;
+    int resample_sample_rate;
     int reformat_pair;
     AVAudioConvert *reformat_ctx;
     AVFifoBuffer *fifo;     /* for compression: one audio fifo per codec */
@@ -768,7 +771,7 @@ static void do_audio_out(AVFormatContext *s,
     int64_t audio_out_size, audio_buf_size;
     int64_t allocated_for_size= size;
 
-    int size_out, frame_bytes, ret;
+    int size_out, frame_bytes, ret, resample_changed;
     AVCodecContext *enc= ost->st->codec;
     AVCodecContext *dec= ist->st->codec;
     int osize= av_get_bits_per_sample_fmt(enc->sample_fmt)/8;
@@ -802,7 +805,28 @@ need_realloc:
     if (enc->channels != dec->channels)
         ost->audio_resample = 1;
 
-    if (ost->audio_resample && !ost->resample) {
+    resample_changed = ost->resample_sample_fmt  != dec->sample_fmt ||
+                       ost->resample_channels    != dec->channels   ||
+                       ost->resample_sample_rate != dec->sample_rate;
+
+    if ((ost->audio_resample && !ost->resample) || resample_changed) {
+        if (resample_changed) {
+            av_log(NULL, AV_LOG_INFO, "Input stream #%d.%d frame changed from rate:%d fmt:%s ch:%d to rate:%d fmt:%s ch:%d\n",
+                   ist->file_index, ist->index,
+                   ost->resample_sample_rate, av_get_sample_fmt_name(ost->resample_sample_fmt), ost->resample_channels,
+                   dec->sample_rate, av_get_sample_fmt_name(dec->sample_fmt), dec->channels);
+            ost->resample_sample_fmt  = dec->sample_fmt;
+            ost->resample_channels    = dec->channels;
+            ost->resample_sample_rate = dec->sample_rate;
+            if (ost->resample)
+                audio_resample_close(ost->resample);
+        }
+        if (ost->resample_sample_fmt  == enc->sample_fmt &&
+            ost->resample_channels    == enc->channels   &&
+            ost->resample_sample_rate == enc->sample_rate) {
+            ost->resample = NULL;
+            ost->audio_resample = 0;
+        } else {
         if (dec->sample_fmt != AV_SAMPLE_FMT_S16)
             fprintf(stderr, "Warning, using s16 intermediate sample format for resampling\n");
         ost->resample = av_audio_resample_init(enc->channels,    dec->channels,
@@ -815,6 +839,7 @@ need_realloc:
                     enc->channels, enc->sample_rate);
             ffmpeg_exit(1);
         }
+        }
     }
 
 #define MAKE_SFMT_PAIR(a,b) ((a)+AV_SAMPLE_FMT_NB*(b))
@@ -2174,6 +2199,9 @@ static int transcode(AVFormatContext **output_files,
                 icodec->request_channels = codec->channels;
                 ist->decoding_needed = 1;
                 ost->encoding_needed = 1;
+                ost->resample_sample_fmt  = icodec->sample_fmt;
+                ost->resample_sample_rate = icodec->sample_rate;
+                ost->resample_channels    = icodec->channels;
                 break;
             case AVMEDIA_TYPE_VIDEO:
                 if (ost->st->codec->pix_fmt == PIX_FMT_NONE) {

-- 
Libav/FFmpeg packaging