diff options
| -rw-r--r-- | ffmpeg.c | 24 | ||||
| -rw-r--r-- | tests/fate2.mak | 2 | ||||
| -rwxr-xr-x | tests/lavf-regression.sh | 12 | ||||
| -rwxr-xr-x | tests/regression-funcs.sh | 2 |
4 files changed, 21 insertions, 19 deletions
@@ -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() |