On 06/15/2011 03:10 PM, Anton Khirnov wrote:
> It's broken with multiple files or audio streams.
>
> This removes the default samplerate of 44100 for raw input, hence all
> the FATE changes.
> ---
> Arrrgh, I could swear I fixed it the first time.
> ---
> 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 04672cc..1a00bdb 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 6a9448f..066f9ef 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 $(SAMPLES)/ansi/TRE-IOM5.ANS -pix_fmt
> rgb24
>
> FATE_TESTS += fate-wmv8-drm
> # discard last packet to avoid fails due to overread of VC-1 decoder
> diff --git a/tests/lavf-regression.sh b/tests/lavf-regression.sh
> index 94d2583..39e752b 100755
> --- a/tests/lavf-regression.sh
> +++ b/tests/lavf-regression.sh
> @@ -14,7 +14,7 @@ eval do_$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 "$do_au" ] ; then
> diff --git a/tests/regression-funcs.sh b/tests/regression-funcs.sh
> index 4cf2e20..e57cdf1 100755
> --- a/tests/regression-funcs.sh
> +++ b/tests/regression-funcs.sh
> @@ -114,7 +114,7 @@ do_video_encoding()
> do_audio_encoding()
> {
> file=${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()
Ok.
-Justin
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