From d7ee44024c96ebdbcd718885a77e9a07779df54c Mon Sep 17 00:00:00 2001 From: Anton Khirnov Date: Wed, 15 Jun 2011 08:00:03 +0200 Subject: 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 04672cc831..1a00bdbb5b 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 6a9448faf1..066f9ef583 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 $(SAMPLES)/ansi/TRE-IOM5.ANS -pix_fmt rgb24 FATE_TESTS += fate-wmv8-drm # discard last packet to avoid fails due to overread of VC-1 decoder diff --git a/tests/lavf-regression.sh b/tests/lavf-regression.sh index 94d258334b..39e752b3c6 100755 --- a/tests/lavf-regression.sh +++ b/tests/lavf-regression.sh @@ -14,7 +14,7 @@ eval do_$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 "$do_au" ] ; then diff --git a/tests/regression-funcs.sh b/tests/regression-funcs.sh index 4cf2e20fd8..e57cdf111e 100755 --- a/tests/regression-funcs.sh +++ b/tests/regression-funcs.sh @@ -114,7 +114,7 @@ do_video_encoding() do_audio_encoding() { file=${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() -- cgit v1.2.3