Andreas Rheinhardt: > Signed-off-by: Andreas Rheinhardt <andreas.rheinha...@outlook.com> > --- > libavformat/aacdec.c | 2 +- > libavformat/adxdec.c | 2 +- > libavformat/dfpwmdec.c | 2 +- > libavformat/gsmdec.c | 2 +- > libavformat/loasdec.c | 2 +- > libavformat/serdec.c | 2 +- > libavformat/wsddec.c | 2 +- > 7 files changed, 7 insertions(+), 7 deletions(-) > > diff --git a/libavformat/aacdec.c b/libavformat/aacdec.c > index 41c9a36239..4da98a6884 100644 > --- a/libavformat/aacdec.c > +++ b/libavformat/aacdec.c > @@ -113,7 +113,7 @@ static int adts_aac_read_header(AVFormatContext *s) > return AVERROR(ENOMEM); > > st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO; > - st->codecpar->codec_id = s->iformat->raw_codec_id; > + st->codecpar->codec_id = AV_CODEC_ID_AAC; > ffstream(st)->need_parsing = AVSTREAM_PARSE_FULL_RAW; > > ff_id3v1_read(s); > diff --git a/libavformat/adxdec.c b/libavformat/adxdec.c > index d808adbf3b..b6bd3303a7 100644 > --- a/libavformat/adxdec.c > +++ b/libavformat/adxdec.c > @@ -120,7 +120,7 @@ static int adx_read_header(AVFormatContext *s) > > par->ch_layout.nb_channels = channels; > par->codec_type = AVMEDIA_TYPE_AUDIO; > - par->codec_id = s->iformat->raw_codec_id; > + par->codec_id = AV_CODEC_ID_ADPCM_ADX; > par->bit_rate = (int64_t)par->sample_rate * > par->ch_layout.nb_channels * BLOCK_SIZE * 8LL / BLOCK_SAMPLES; > > avpriv_set_pts_info(st, 64, BLOCK_SAMPLES, par->sample_rate); > diff --git a/libavformat/dfpwmdec.c b/libavformat/dfpwmdec.c > index 685b95148c..b92b00f13a 100644 > --- a/libavformat/dfpwmdec.c > +++ b/libavformat/dfpwmdec.c > @@ -50,7 +50,7 @@ static int dfpwm_read_header(AVFormatContext *s) > par = st->codecpar; > > par->codec_type = AVMEDIA_TYPE_AUDIO; > - par->codec_id = s->iformat->raw_codec_id; > + par->codec_id = AV_CODEC_ID_DFPWM; > par->sample_rate = s1->sample_rate; > #if FF_API_OLD_CHANNEL_LAYOUT > if (s1->ch_layout.nb_channels) { > diff --git a/libavformat/gsmdec.c b/libavformat/gsmdec.c > index 09dc0e0fb3..7150daa510 100644 > --- a/libavformat/gsmdec.c > +++ b/libavformat/gsmdec.c > @@ -78,7 +78,7 @@ static int gsm_read_header(AVFormatContext *s) > return AVERROR(ENOMEM); > > st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO; > - st->codecpar->codec_id = s->iformat->raw_codec_id; > + st->codecpar->codec_id = AV_CODEC_ID_GSM; > st->codecpar->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO; > st->codecpar->sample_rate = c->sample_rate; > st->codecpar->bit_rate = GSM_BLOCK_SIZE * 8 * c->sample_rate / > GSM_BLOCK_SAMPLES; > diff --git a/libavformat/loasdec.c b/libavformat/loasdec.c > index e739b6c196..7b8b2ea4bc 100644 > --- a/libavformat/loasdec.c > +++ b/libavformat/loasdec.c > @@ -74,7 +74,7 @@ static int loas_read_header(AVFormatContext *s) > return AVERROR(ENOMEM); > > st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO; > - st->codecpar->codec_id = s->iformat->raw_codec_id; > + st->codecpar->codec_id = AV_CODEC_ID_AAC_LATM; > ffstream(st)->need_parsing = AVSTREAM_PARSE_FULL_RAW; > > //LCM of all possible AAC sample rates > diff --git a/libavformat/serdec.c b/libavformat/serdec.c > index 11add35b32..639c899249 100644 > --- a/libavformat/serdec.c > +++ b/libavformat/serdec.c > @@ -80,7 +80,7 @@ static int ser_read_header(AVFormatContext *s) > } > > st->codecpar->codec_type = AVMEDIA_TYPE_VIDEO; > - st->codecpar->codec_id = s->iformat->raw_codec_id; > + st->codecpar->codec_id = AV_CODEC_ID_RAWVIDEO; > > avpriv_set_pts_info(st, 64, ser->framerate.den, ser->framerate.num); > > diff --git a/libavformat/wsddec.c b/libavformat/wsddec.c > index 9bee4d51bb..8153d898dd 100644 > --- a/libavformat/wsddec.c > +++ b/libavformat/wsddec.c > @@ -125,7 +125,7 @@ static int wsd_read_header(AVFormatContext *s) > av_dict_set(&s->metadata, "playback_time", playback_time, 0); > > st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO; > - st->codecpar->codec_id = s->iformat->raw_codec_id; > + st->codecpar->codec_id = AV_CODEC_ID_DSD_MSBF; > st->codecpar->sample_rate = avio_rb32(pb) / 8; > avio_skip(pb, 4); > st->codecpar->ch_layout.nb_channels = avio_r8(pb) & 0xF;
Will apply this tomorrow unless there are objections. - Andreas _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-requ...@ffmpeg.org with subject "unsubscribe".