On Sat, Nov 27, 2021 at 04:45:51PM -0500, John-Paul Stewart wrote:
> Adds support for reading audio sample size from the data instead of
> assuming all audio is 16 bits per sample.
> ---
> libavformat/mvdec.c | 18 +++++++++++++++---
> 1 file changed, 15 insertions(+), 3 deletions(-)
>
> diff --git a/libavformat/mvdec.c b/libavformat/mvdec.c
> index 8492928820..8b54a9ab04 100644
> --- a/libavformat/mvdec.c
> +++ b/libavformat/mvdec.c
> @@ -299,6 +299,8 @@ static int mv_read_header(AVFormatContext *avctx)
> if (version == 2) {
> uint64_t timestamp;
> int v;
> + uint32_t bytes_per_sample;
> +
> avio_skip(pb, 22);
>
> /* allocate audio track first to prevent unnecessary seeking
> @@ -341,11 +343,21 @@ static int mv_read_header(AVFormatContext *avctx)
> }
> avpriv_set_pts_info(ast, 33, 1, ast->codecpar->sample_rate);
>
> - avio_skip(pb, 4);
> + bytes_per_sample = avio_rb32(pb);
>
> v = avio_rb32(pb);
> if (v == AUDIO_FORMAT_SIGNED) {
> - ast->codecpar->codec_id = AV_CODEC_ID_PCM_S16BE;
> + switch (bytes_per_sample) {
> + case 1:
> + ast->codecpar->codec_id = AV_CODEC_ID_PCM_S8;
> + break;
> + case 2:
> + ast->codecpar->codec_id = AV_CODEC_ID_PCM_S16BE;
> + break;
> + default:
> + avpriv_request_sample(avctx, "Audio sample size %i bytes",
> bytes_per_sample);
> + break;
> + }
> } else {
> avpriv_request_sample(avctx, "Audio compression (format %i)", v);
> }
> @@ -369,7 +381,7 @@ static int mv_read_header(AVFormatContext *avctx)
> avio_skip(pb, 8);
> av_add_index_entry(ast, pos, timestamp, asize, 0,
> AVINDEX_KEYFRAME);
> av_add_index_entry(vst, pos + asize, i, vsize, 0,
> AVINDEX_KEYFRAME);
> - timestamp += asize / (ast->codecpar->channels * 2LL);
> + timestamp += asize / (ast->codecpar->channels *
> bytes_per_sample);
> }
> } else if (!version && avio_rb16(pb) == 3) {
> avio_skip(pb, 4);
> --
> 2.33.0thanks for doing this. both patches look good. i will apply in a few days. -- Peter (A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B)
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