Zane van Iperen: > These files are technically a series of planar mono tracks. > If the "music" flag is set, merge the packets from the two > mono tracks, essentially replicating: > > [0:a:0][0:a:1]join=inputs=2:channel_layout=stereo[a] > > Signed-off-by: Zane van Iperen <z...@zanevaniperen.com> > --- > libavformat/pp_bnk.c | 60 ++++++++++++++++++++++++++++++++++++-------- > 1 file changed, 50 insertions(+), 10 deletions(-) > > diff --git a/libavformat/pp_bnk.c b/libavformat/pp_bnk.c > index 8364de1fd9..970ef09923 100644 > --- a/libavformat/pp_bnk.c > +++ b/libavformat/pp_bnk.c > @@ -55,6 +55,8 @@ typedef struct PPBnkCtx { > int track_count; > PPBnkCtxTrack *tracks; > uint32_t current_track; > + int is_music; > + AVPacket pkt; > } PPBnkCtx; > > enum { > @@ -194,8 +196,12 @@ static int pp_bnk_read_header(AVFormatContext *s) > goto fail; > } > > + ctx->is_music = (hdr.flags & PP_BNK_FLAG_MUSIC) && > + (ctx->track_count == 2) && > + (ctx->tracks[0].data_size == ctx->tracks[1].data_size); > + > /* Build the streams. */ > - for (int i = 0; i < ctx->track_count; i++) { > + for (int i = 0; i < (ctx->is_music ? 1 : ctx->track_count); i++) { > if (!(st = avformat_new_stream(s, NULL))) { > ret = AVERROR(ENOMEM); > goto fail; > @@ -204,14 +210,21 @@ static int pp_bnk_read_header(AVFormatContext *s) > par = st->codecpar; > par->codec_type = AVMEDIA_TYPE_AUDIO; > par->codec_id = AV_CODEC_ID_ADPCM_IMA_CUNNING; > - par->format = AV_SAMPLE_FMT_S16; > - par->channel_layout = AV_CH_LAYOUT_MONO; > - par->channels = 1; > + par->format = AV_SAMPLE_FMT_S16P; > + > + if (ctx->is_music) { > + par->channel_layout = AV_CH_LAYOUT_STEREO; > + par->channels = 2; > + } else { > + par->channel_layout = AV_CH_LAYOUT_MONO; > + par->channels = 1; > + } > + > par->sample_rate = hdr.sample_rate; > par->bits_per_coded_sample = 4; > par->bits_per_raw_sample = 16; > par->block_align = 1; > - par->bit_rate = par->sample_rate * > par->bits_per_coded_sample; > + par->bit_rate = par->sample_rate * > par->bits_per_coded_sample * par->channels; > > avpriv_set_pts_info(st, 64, 1, par->sample_rate); > st->start_time = 0; > @@ -253,7 +266,22 @@ static int pp_bnk_read_packet(AVFormatContext *s, > AVPacket *pkt) > > size = FFMIN(trk->data_size - trk->bytes_read, PP_BNK_MAX_READ_SIZE); > > - if ((ret = av_get_packet(s->pb, pkt, size)) == AVERROR_EOF) { > + if (!ctx->is_music) > + ret = av_new_packet(&ctx->pkt, size); > + else if (ctx->current_track == 0) > + ret = av_new_packet(&ctx->pkt, size * 2); > + else > + ret = 0; > + > + if (ret < 0) > + return ret; > + > + if (ctx->is_music) > + ret = avio_read(s->pb, ctx->pkt.data + size * > ctx->current_track, size); > + else > + ret = avio_read(s->pb, ctx->pkt.data, size); > + > + if (ret == AVERROR_EOF) { > /* If we've hit EOF, don't attempt this track again. */ > trk->data_size = trk->bytes_read; > continue; > @@ -261,10 +289,21 @@ static int pp_bnk_read_packet(AVFormatContext *s, > AVPacket *pkt) > return ret; > } > > - trk->bytes_read += ret; > - pkt->flags &= ~AV_PKT_FLAG_CORRUPT; > - pkt->stream_index = ctx->current_track++; > - pkt->duration = ret * 2; > + trk->bytes_read += ret; > + ctx->pkt.flags &= ~AV_PKT_FLAG_CORRUPT; > + ctx->pkt.stream_index = ctx->current_track++; > + ctx->pkt.duration = ret * 2; > + > + if (ctx->is_music) { > + if (ctx->pkt.stream_index == 0) > + return FFERROR_REDO;
Wouldn't a simple continue have the same effect? This would allow to avoid the temporary packet. > + > + ctx->pkt.stream_index = 0; > + } else { > + ctx->pkt.size = ret; > + } > + > + av_packet_move_ref(pkt, &ctx->pkt); > return 0; > } > > @@ -277,6 +316,7 @@ static int pp_bnk_read_close(AVFormatContext *s) > PPBnkCtx *ctx = s->priv_data; > > av_freep(&ctx->tracks); > + av_packet_unref(&ctx->pkt); > > return 0; > } > _______________________________________________ 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".