Paul B Mahol:
> Why you put nonsense regression part in log.
>
> That is very unfriendly behavior from you.
>
> Noted.
>
Before your patch, the audio in the files in question would work; with
your patch it doesn't any longer. That's a regression.
> 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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>>
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>> ffmpeg-devel-requ...@ffmpeg.org with subject "unsubscribe".
>>
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