Why you put nonsense regression part in log.
That is very unfriendly behavior from you.
Noted.
On Sun, Apr 18, 2021 at 1:50 AM Andreas Rheinhardt <
andreas.rheinha...@outlook.com> wrote:
> Andreas Rheinhardt:
> > Up until now the cover images will get the stream index 0 in this case,
> > violating the hardcoded assumption that this is the index of the audio
> > stream. Fix this by creating the audio stream first; this is also in
> > line with the expectations of ff_pcm_read_seek() and
> > ff_spdif_read_packet(). It also simplifies the code to parse the fmt and
> > xma2 tags.
> >
> > Fixes #8540; regression since f5aad350d3695b5b16e7d135154a4c61e4dce9d8.
> >
> > Signed-off-by: Andreas Rheinhardt <andreas.rheinha...@outlook.com>
> > ---
> > libavformat/wavdec.c | 78 ++++++++++++++++++++++----------------------
> > 1 file changed, 39 insertions(+), 39 deletions(-)
> >
> > diff --git a/libavformat/wavdec.c b/libavformat/wavdec.c
> > index 8214ab8498..791ae23b4a 100644
> > --- a/libavformat/wavdec.c
> > +++ b/libavformat/wavdec.c
> > @@ -49,6 +49,7 @@ typedef struct WAVDemuxContext {
> > const AVClass *class;
> > int64_t data_end;
> > int w64;
> > + AVStream *vst;
> > int64_t smv_data_ofs;
> > int smv_block_size;
> > int smv_frames_per_jpeg;
> > @@ -170,30 +171,26 @@ static void handle_stream_probing(AVStream *st)
> > }
> > }
> >
> > -static int wav_parse_fmt_tag(AVFormatContext *s, int64_t size, AVStream
> **st)
> > +static int wav_parse_fmt_tag(AVFormatContext *s, int64_t size, AVStream
> *st)
> > {
> > AVIOContext *pb = s->pb;
> > WAVDemuxContext *wav = s->priv_data;
> > int ret;
> >
> > /* parse fmt header */
> > - *st = avformat_new_stream(s, NULL);
> > - if (!*st)
> > - return AVERROR(ENOMEM);
> > -
> > - ret = ff_get_wav_header(s, pb, (*st)->codecpar, size, wav->rifx);
> > + ret = ff_get_wav_header(s, pb, st->codecpar, size, wav->rifx);
> > if (ret < 0)
> > return ret;
> > - handle_stream_probing(*st);
> > + handle_stream_probing(st);
> >
> > - (*st)->need_parsing = AVSTREAM_PARSE_FULL_RAW;
> > + st->need_parsing = AVSTREAM_PARSE_FULL_RAW;
> >
> > - avpriv_set_pts_info(*st, 64, 1, (*st)->codecpar->sample_rate);
> > + avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
> >
> > return 0;
> > }
> >
> > -static int wav_parse_xma2_tag(AVFormatContext *s, int64_t size,
> AVStream **st)
> > +static int wav_parse_xma2_tag(AVFormatContext *s, int64_t size,
> AVStream *st)
> > {
> > AVIOContext *pb = s->pb;
> > int version, num_streams, i, channels = 0, ret;
> > @@ -201,13 +198,9 @@ static int wav_parse_xma2_tag(AVFormatContext *s,
> int64_t size, AVStream **st)
> > if (size < 36)
> > return AVERROR_INVALIDDATA;
> >
> > - *st = avformat_new_stream(s, NULL);
> > - if (!*st)
> > - return AVERROR(ENOMEM);
> > -
> > - (*st)->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
> > - (*st)->codecpar->codec_id = AV_CODEC_ID_XMA2;
> > - (*st)->need_parsing = AVSTREAM_PARSE_FULL_RAW;
> > + st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
> > + st->codecpar->codec_id = AV_CODEC_ID_XMA2;
> > + st->need_parsing = AVSTREAM_PARSE_FULL_RAW;
> >
> > version = avio_r8(pb);
> > if (version != 3 && version != 4)
> > @@ -216,26 +209,26 @@ static int wav_parse_xma2_tag(AVFormatContext *s,
> int64_t size, AVStream **st)
> > if (size != (32 + ((version==3)?0:8) + 4*num_streams))
> > return AVERROR_INVALIDDATA;
> > avio_skip(pb, 10);
> > - (*st)->codecpar->sample_rate = avio_rb32(pb);
> > + st->codecpar->sample_rate = avio_rb32(pb);
> > if (version == 4)
> > avio_skip(pb, 8);
> > avio_skip(pb, 4);
> > - (*st)->duration = avio_rb32(pb);
> > + st->duration = avio_rb32(pb);
> > avio_skip(pb, 8);
> >
> > for (i = 0; i < num_streams; i++) {
> > channels += avio_r8(pb);
> > avio_skip(pb, 3);
> > }
> > - (*st)->codecpar->channels = channels;
> > + st->codecpar->channels = channels;
> >
> > - if ((*st)->codecpar->channels <= 0 || (*st)->codecpar->sample_rate
> <= 0)
> > + if (st->codecpar->channels <= 0 || st->codecpar->sample_rate <= 0)
> > return AVERROR_INVALIDDATA;
> >
> > - avpriv_set_pts_info(*st, 64, 1, (*st)->codecpar->sample_rate);
> > + avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
> >
> > avio_seek(pb, -size, SEEK_CUR);
> > - if ((ret = ff_get_extradata(s, (*st)->codecpar, pb, size)) < 0)
> > + if ((ret = ff_get_extradata(s, st->codecpar, pb, size)) < 0)
> > return ret;
> >
> > return 0;
> > @@ -407,6 +400,11 @@ static int wav_read_header(AVFormatContext *s)
> >
> > }
> >
> > + /* Create the audio stream now so that its index is always zero */
> > + st = avformat_new_stream(s, NULL);
> > + if (!st)
> > + return AVERROR(ENOMEM);
> > +
> > for (;;) {
> > AVStream *vst;
> > size = next_tag(pb, &tag, wav->rifx);
> > @@ -418,7 +416,7 @@ static int wav_read_header(AVFormatContext *s)
> > switch (tag) {
> > case MKTAG('f', 'm', 't', ' '):
> > /* only parse the first 'fmt ' tag found */
> > - if (!got_xma2 && !got_fmt && (ret = wav_parse_fmt_tag(s,
> size, &st)) < 0) {
> > + if (!got_xma2 && !got_fmt && (ret = wav_parse_fmt_tag(s,
> size, st)) < 0) {
> > return ret;
> > } else if (got_fmt)
> > av_log(s, AV_LOG_WARNING, "found more than one 'fmt '
> tag\n");
> > @@ -427,7 +425,7 @@ static int wav_read_header(AVFormatContext *s)
> > break;
> > case MKTAG('X', 'M', 'A', '2'):
> > /* only parse the first 'XMA2' tag found */
> > - if (!got_fmt && !got_xma2 && (ret = wav_parse_xma2_tag(s,
> size, &st)) < 0) {
> > + if (!got_fmt && !got_xma2 && (ret = wav_parse_xma2_tag(s,
> size, st)) < 0) {
> > return ret;
> > } else if (got_xma2)
> > av_log(s, AV_LOG_WARNING, "found more than one 'XMA2'
> tag\n");
> > @@ -484,6 +482,7 @@ static int wav_read_header(AVFormatContext *s)
> > vst = avformat_new_stream(s, NULL);
> > if (!vst)
> > return AVERROR(ENOMEM);
> > + wav->vst = vst;
> > avio_r8(pb);
> > vst->id = 1;
> > vst->codecpar->codec_type = AVMEDIA_TYPE_VIDEO;
> > @@ -693,23 +692,24 @@ static int wav_read_packet(AVFormatContext *s,
> AVPacket *pkt)
> > {
> > int ret, size;
> > int64_t left;
> > - AVStream *st;
> > WAVDemuxContext *wav = s->priv_data;
> > + AVStream *st = s->streams[0];
> >
> > if (CONFIG_SPDIF_DEMUXER && wav->spdif == 1)
> > return ff_spdif_read_packet(s, pkt);
> >
> > if (wav->smv_data_ofs > 0) {
> > int64_t audio_dts, video_dts;
> > + AVStream *vst = wav->vst;
> > smv_retry:
> > - audio_dts = (int32_t)s->streams[0]->cur_dts;
> > - video_dts = (int32_t)s->streams[1]->cur_dts;
> > + audio_dts = (int32_t)st->cur_dts;
> > + video_dts = (int32_t)vst->cur_dts;
> >
> > if (audio_dts != AV_NOPTS_VALUE && video_dts != AV_NOPTS_VALUE)
> {
> > /*We always return a video frame first to get the pixel
> format first*/
> > wav->smv_last_stream = wav->smv_given_first ?
> > - av_compare_ts(video_dts, s->streams[1]->time_base,
> > - audio_dts, s->streams[0]->time_base) > 0
> : 0;
> > + av_compare_ts(video_dts, vst->time_base,
> > + audio_dts, st->time_base) > 0 : 0;
> > wav->smv_given_first = 1;
> > }
> > wav->smv_last_stream = !wav->smv_last_stream;
> > @@ -732,7 +732,7 @@ smv_retry:
> > pkt->duration = wav->smv_frames_per_jpeg;
> > wav->smv_block++;
> >
> > - pkt->stream_index = 1;
> > + pkt->stream_index = vst->index;
> > smv_out:
> > avio_seek(s->pb, old_pos, SEEK_SET);
> > if (ret == AVERROR_EOF) {
> > @@ -743,8 +743,6 @@ smv_out:
> > }
> > }
> >
> > - st = s->streams[0];
> > -
> > left = wav->data_end - avio_tell(s->pb);
> > if (wav->ignore_length)
> > left = INT_MAX;
> > @@ -781,22 +779,24 @@ static int wav_read_seek(AVFormatContext *s,
> > int stream_index, int64_t timestamp, int flags)
> > {
> > WAVDemuxContext *wav = s->priv_data;
> > - AVStream *st;
> > + AVStream *ast = s->streams[0], *vst = wav->vst;
> > wav->smv_eof = 0;
> > wav->audio_eof = 0;
> > +
> > + if (stream_index != 0 && (!vst || stream_index != vst->index))
> > + return AVERROR(EINVAL);
> > if (wav->smv_data_ofs > 0) {
> > int64_t smv_timestamp = timestamp;
> > if (stream_index == 0)
> > - smv_timestamp = av_rescale_q(timestamp,
> s->streams[0]->time_base, s->streams[1]->time_base);
> > + smv_timestamp = av_rescale_q(timestamp, ast->time_base,
> vst->time_base);
> > else
> > - timestamp = av_rescale_q(smv_timestamp,
> s->streams[1]->time_base, s->streams[0]->time_base);
> > + timestamp = av_rescale_q(smv_timestamp, vst->time_base,
> ast->time_base);
> > if (wav->smv_frames_per_jpeg > 0) {
> > wav->smv_block = smv_timestamp / wav->smv_frames_per_jpeg;
> > }
> > }
> >
> > - st = s->streams[0];
> > - switch (st->codecpar->codec_id) {
> > + switch (ast->codecpar->codec_id) {
> > case AV_CODEC_ID_MP2:
> > case AV_CODEC_ID_MP3:
> > case AV_CODEC_ID_AC3:
> > @@ -807,7 +807,7 @@ static int wav_read_seek(AVFormatContext *s,
> > default:
> > break;
> > }
> > - return ff_pcm_read_seek(s, stream_index, timestamp, flags);
> > + return ff_pcm_read_seek(s, 0, timestamp, flags);
> > }
> >
> > static const AVClass wav_demuxer_class = {
> >
> Will apply unless there are objections.
>
> - Andreas
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