har0ke/FFmpeg
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

avformat/mvdec: re-indent after last commit

Browse Source 
Signed-off-by: John-Paul Stewart <jpstewart@personalprojects.net>
Reviewed-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Reviewed-by: Peter Ross <pross@xvid.org>
This commit is contained in:
John-Paul Stewart 2021-12-31 20:11:48 -05:00 committed by Peter Ross
parent a8da115143
commit b24f0c82b3
1 changed files with 36 additions and 36 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

72
libavformat/mvdec.c
View File

@ -316,9 +316,9 @@ static int mv_read_header(AVFormatContext *avctx)
v = avio_rb16(pb);
if (v == MOVIE_SOUND) {
/* movie has sound so allocate an audio stream */
ast = avformat_new_stream(avctx, NULL);
if (!ast)
return AVERROR(ENOMEM);
ast = avformat_new_stream(avctx, NULL);
if (!ast)
return AVERROR(ENOMEM);
} else if (v != MOVIE_SILENT)
return AVERROR_INVALIDDATA;
@ -350,41 +350,41 @@ static int mv_read_header(AVFormatContext *avctx)
avio_skip(pb, 12);
if (ast) {
ast->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
ast->nb_frames = vst->nb_frames;
ast->codecpar->sample_rate = avio_rb32(pb);
if (ast->codecpar->sample_rate <= 0) {
av_log(avctx, AV_LOG_ERROR, "Invalid sample rate %d\n", ast->codecpar->sample_rate);
return AVERROR_INVALIDDATA;
}
avpriv_set_pts_info(ast, 33, 1, ast->codecpar->sample_rate);
bytes_per_sample = avio_rb32(pb);
v = avio_rb32(pb);
if (v == AUDIO_FORMAT_SIGNED) {
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;
ast->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
ast->nb_frames = vst->nb_frames;
ast->codecpar->sample_rate = avio_rb32(pb);
if (ast->codecpar->sample_rate <= 0) {
av_log(avctx, AV_LOG_ERROR, "Invalid sample rate %d\n", ast->codecpar->sample_rate);
return AVERROR_INVALIDDATA;
}
} else {
avpriv_request_sample(avctx, "Audio compression (format %i)", v);
}
avpriv_set_pts_info(ast, 33, 1, ast->codecpar->sample_rate);
if (bytes_per_sample == 0)
return AVERROR_INVALIDDATA;
bytes_per_sample = avio_rb32(pb);
if (set_channels(avctx, ast, avio_rb32(pb)) < 0)
return AVERROR_INVALIDDATA;
v = avio_rb32(pb);
if (v == AUDIO_FORMAT_SIGNED) {
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);
}
avio_skip(pb, 8);
if (bytes_per_sample == 0)
return AVERROR_INVALIDDATA;
if (set_channels(avctx, ast, avio_rb32(pb)) < 0)
return AVERROR_INVALIDDATA;
avio_skip(pb, 8);
} else
avio_skip(pb, 24); /* skip meaningless audio metadata */
@ -401,8 +401,8 @@ static int mv_read_header(AVFormatContext *avctx)
return AVERROR_INVALIDDATA;
avio_skip(pb, 8);
if (ast) {
av_add_index_entry(ast, pos, timestamp, asize, 0, AVINDEX_KEYFRAME);
timestamp += asize / (ast->codecpar->channels * (uint64_t)bytes_per_sample);
av_add_index_entry(ast, pos, timestamp, asize, 0, AVINDEX_KEYFRAME);
timestamp += asize / (ast->codecpar->channels * (uint64_t)bytes_per_sample);
}
av_add_index_entry(vst, pos + asize, i, vsize, 0, AVINDEX_KEYFRAME);
}