From d7ee44024c96ebdbcd718885a77e9a07779df54c Mon Sep 17 00:00:00 2001
From: Anton Khirnov <anton@khirnov.net>
Date: Wed, 15 Jun 2011 08:00:03 +0200
Subject: [PATCH] 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 | 24 +++++++++++++-----------
tests/fate2.mak | 2 +-
tests/lavf-regression.sh | 12 ++++++------
tests/regression-funcs.sh | 2 +-
4 files changed, 21 insertions(+), 19 deletions(-)
diff --git a/ffmpeg.c b/ffmpeg.c
index 04672cc8316..1a00bdbb5ba 100644
--- a/ffmpeg.c
+++ b/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();
diff --git a/tests/fate2.mak b/tests/fate2.mak
index 6a9448faf16..066f9ef5837 100644
--- a/tests/fate2.mak
+++ b/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 $(SAMPLES)/ansi/TRE-IOM5.ANS -pix_fmt rgb24
FATE_TESTS += fate-wmv8-drm
# discard last packet to avoid fails due to overread of VC-1 decoder
diff --git a/tests/lavf-regression.sh b/tests/lavf-regression.sh
index 94d258334bf..39e752b3c67 100755
--- a/tests/lavf-regression.sh
+++ b/tests/lavf-regression.sh
@@ -14,7 +14,7 @@ eval do_$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 "$do_au" ] ; then
diff --git a/tests/regression-funcs.sh b/tests/regression-funcs.sh
index 4cf2e20fd80..e57cdf111ec 100755
--- a/tests/regression-funcs.sh
+++ b/tests/regression-funcs.sh
@@ -114,7 +114,7 @@ do_video_encoding()
do_audio_encoding()
{
file=${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()
--
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