| Message ID | HE1PR0301MB215469B69C9D56EFC4F0A1A78F4B9@HE1PR0301MB2154.eurprd03.prod.outlook.com |
|---|---|
| State | Accepted |
| Commit | e1f41b4fdc5d3e8813d8f9d4d81899bd4e6df31b |
| Headers | show |
| Series | [FFmpeg-devel,v2] avformat/wavdec: Fix reading files with id3v2 apic before fmt tag | expand |
| Context | Check | Description |
|---|---|---|
| andriy/x86_make | success | Make finished |
| andriy/x86_make_fate | success | Make fate finished |
| andriy/PPC64_make | success | Make finished |
| andriy/PPC64_make_fate | success | Make fate finished |
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.rheinhardt@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
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.rheinhardt@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.rheinhardt@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-request@ffmpeg.org with subject "unsubscribe". >
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.rheinhardt@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.rheinhardt@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-request@ffmpeg.org with subject "unsubscribe". >> > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > https://ffmpeg.org/mailman/listinfo/ffmpeg-devel > > To unsubscribe, visit link above, or email > ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". >
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 = {
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.rheinhardt@outlook.com> --- libavformat/wavdec.c | 78 ++++++++++++++++++++++---------------------- 1 file changed, 39 insertions(+), 39 deletions(-)