ffmpeg: don't abuse a global for passing samplerate from input to output

It's broken with multiple files or audio streams.

This removes the default samplerate of 44100 for raw input, hence all
the FATE changes.

ffmpeg.c

@@ -163,7 +163,7 @@ static char *vfilters = NULL;
 #endif
 
 static int intra_only = 0;
-static int audio_sample_rate = 44100;
+static int audio_sample_rate = 0;
 static int64_t channel_layout = 0;
 #define QSCALE_NONE -99999
 static float audio_qscale = QSCALE_NONE;
@@ -2170,6 +2170,13 @@ static int transcode(AVFormatContext **output_files,
                 if(!ost->fifo)
                     goto fail;
                 ost->reformat_pair = MAKE_SFMT_PAIR(AV_SAMPLE_FMT_NONE,AV_SAMPLE_FMT_NONE);
+                if (!codec->sample_rate) {
+                    codec->sample_rate = icodec->sample_rate;
+                    if (icodec->lowres)
+                        codec->sample_rate >>= icodec->lowres;
+                }
+                choose_sample_rate(ost->st, codec->codec);
+                codec->time_base = (AVRational){1, codec->sample_rate};
                 ost->audio_resample = codec->sample_rate != icodec->sample_rate || audio_sync_method > 1;
                 icodec->request_channels = codec->channels;
                 ist->decoding_needed = 1;
@@ -3268,15 +3275,9 @@ static int opt_input_file(const char *opt, const char *filename)
             set_context_opts(dec, avcodec_opts[AVMEDIA_TYPE_AUDIO], AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM, input_codecs[nb_input_codecs-1]);
             channel_layout    = dec->channel_layout;
             audio_channels    = dec->channels;
-            audio_sample_rate = dec->sample_rate;
             audio_sample_fmt  = dec->sample_fmt;
             if(audio_disable)
                 st->discard= AVDISCARD_ALL;
-            /* Note that av_find_stream_info can add more streams, and we
-             * currently have no chance of setting up lowres decoding
-             * early enough for them. */
-            if (dec->lowres)
-                audio_sample_rate >>= dec->lowres;
             break;
         case AVMEDIA_TYPE_VIDEO:
             input_codecs[nb_input_codecs-1] = avcodec_find_decoder_by_name(video_codec_name);
@@ -3338,6 +3339,7 @@ static int opt_input_file(const char *opt, const char *filename)
     input_files[nb_input_files - 1].ist_index  = nb_input_streams - ic->nb_streams;
 
     video_channel = 0;
+    audio_sample_rate = 0;
 
     av_freep(&video_codec_name);
     av_freep(&audio_codec_name);
@@ -3585,7 +3587,6 @@ static void new_audio_stream(AVFormatContext *oc, int file_idx)
     if (audio_stream_copy) {
         st->stream_copy = 1;
         audio_enc->channels = audio_channels;
-        audio_enc->sample_rate = audio_sample_rate;
     } else {
         audio_enc->codec_id = codec_id;
         set_context_opts(audio_enc, avcodec_opts[AVMEDIA_TYPE_AUDIO], AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, codec);
@@ -3596,14 +3597,13 @@ static void new_audio_stream(AVFormatContext *oc, int file_idx)
         }
         audio_enc->channels = audio_channels;
         audio_enc->sample_fmt = audio_sample_fmt;
-        audio_enc->sample_rate = audio_sample_rate;
+        if (audio_sample_rate)
+            audio_enc->sample_rate = audio_sample_rate;
         audio_enc->channel_layout = channel_layout;
         if (av_get_channel_layout_nb_channels(channel_layout) != audio_channels)
             audio_enc->channel_layout = 0;
         choose_sample_fmt(st, codec);
-        choose_sample_rate(st, codec);
     }
-    audio_enc->time_base= (AVRational){1, audio_sample_rate};
     if (audio_language) {
         av_dict_set(&st->metadata, "language", audio_language, 0);
         av_freep(&audio_language);
@@ -3889,6 +3889,8 @@ static void opt_output_file(const char *filename)
 
     set_context_opts(oc, avformat_opts, AV_OPT_FLAG_ENCODING_PARAM, NULL);
 
+    audio_sample_rate = 0;
+
     av_freep(&forced_key_frames);
     uninit_opts();
     init_opts();

tests/fate2.mak

@@ -165,7 +165,7 @@ fate-wmapro-2ch: CMP = oneoff
 fate-wmapro-2ch: REF = $(SAMPLES)/wmapro/Beethovens_9th-1_small.pcm
 
 FATE_TESTS += fate-ansi
-fate-ansi: CMD = framecrc -i $(SAMPLES)/ansi/TRE-IOM5.ANS -pix_fmt rgb24
+fate-ansi: CMD = framecrc -ar 44100 -i $(SAMPLES)/ansi/TRE-IOM5.ANS -pix_fmt rgb24
 
 FATE_TESTS += fate-wmv8-drm
 # discard last packet to avoid fails due to overread of VC-1 decoder

tests/lavf-regression.sh

@@ -14,7 +14,7 @@ eval do_$test=y
 do_lavf()
 {
     file=${outfile}lavf.$1
-    do_ffmpeg $file $DEC_OPTS -f image2 -vcodec pgmyuv -i $raw_src $DEC_OPTS -f s16le -i $pcm_src $ENC_OPTS -t 1 -qscale 10 $2
+    do_ffmpeg $file $DEC_OPTS -f image2 -vcodec pgmyuv -i $raw_src $DEC_OPTS -ar 44100 -f s16le -i $pcm_src $ENC_OPTS -t 1 -qscale 10 $2
     do_ffmpeg_crc $file $DEC_OPTS -i $target_path/$file $3
 }
 
@@ -39,8 +39,8 @@ do_image_formats()
 do_audio_only()
 {
     file=${outfile}lavf.$1
-    do_ffmpeg $file $DEC_OPTS $2 -f s16le -i $pcm_src $ENC_OPTS -t 1 -qscale 10 $3
-    do_ffmpeg_crc $file $DEC_OPTS -i $target_path/$file
+    do_ffmpeg $file $DEC_OPTS $2 -ar 44100 -f s16le -i $pcm_src $ENC_OPTS -t 1 -qscale 10 $3
+    do_ffmpeg_crc $file $DEC_OPTS $4 -i $target_path/$file
 }
 
 rm -f "$logfile"
@@ -55,7 +55,7 @@ fi
 
 if [ -n "$do_rm" ] ; then
 file=${outfile}lavf.rm
-do_ffmpeg $file $DEC_OPTS -f image2 -vcodec pgmyuv -i $raw_src $DEC_OPTS -f s16le -i $pcm_src $ENC_OPTS -t 1 -qscale 10 -acodec ac3_fixed
+do_ffmpeg $file $DEC_OPTS -f image2 -vcodec pgmyuv -i $raw_src $DEC_OPTS -ar 44100 -f s16le -i $pcm_src $ENC_OPTS -t 1 -qscale 10 -acodec ac3_fixed
 # broken
 #do_ffmpeg_crc $file -i $target_path/$file
 fi
@@ -181,11 +181,11 @@ do_audio_only wav
 fi
 
 if [ -n "$do_alaw" ] ; then
-do_audio_only al
+do_audio_only al "" "" "-ar 44100"
 fi
 
 if [ -n "$do_mulaw" ] ; then
-do_audio_only ul
+do_audio_only ul "" "" "-ar 44100"
 fi
 
 if [ -n "$do_au" ] ; then

tests/regression-funcs.sh

@@ -114,7 +114,7 @@ do_video_encoding()
 do_audio_encoding()
 {
     file=${outfile}$1
-    do_ffmpeg $file $DEC_OPTS -ac 2 -f s16le -i $pcm_src -ab 128k $ENC_OPTS $2
+    do_ffmpeg $file $DEC_OPTS -ac 2 -ar 44100 -f s16le -i $pcm_src -ab 128k $ENC_OPTS $2
 }
 
 do_audio_decoding()