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
_______________________________________________
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".
