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authorAnton Khirnov <anton@khirnov.net>2011-06-15 08:00:03 +0200
committerAnton Khirnov <anton@khirnov.net>2011-06-15 21:57:52 +0200
commitd7ee44024c96ebdbcd718885a77e9a07779df54c (patch)
treeacef5df8b963857d97380d51fe5dfb7bca094abd /ffmpeg.c
parent88ff180ad66d5b12f5ee0ffbda891b467725a8d3 (diff)
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.
Diffstat (limited to 'ffmpeg.c')
-rw-r--r--ffmpeg.c24
1 files changed, 13 insertions, 11 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();