[FFmpeg-cvslog] avdevice/oss_dec: account for sample size when computing timestamp
Matt Jacobson
git at videolan.org
Mon Jun 20 00:03:09 EEST 2022
ffmpeg | branch: master | Matt Jacobson <mhjacobson at me.com> | Wed Jun 1 05:06:16 2022 -0400| [b3e261bab381f43ab5f842725d30479d511d1111] | committer: Marton Balint
avdevice/oss_dec: account for sample size when computing timestamp
Don't assume each sample is one byte in size. Doing so results in wrong and
occasionally non-monotonically-increasing timestamps.
Fix nearby cosmetic typo.
Signed-off-by: Marton Balint <cus at passwd.hu>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=b3e261bab381f43ab5f842725d30479d511d1111
---
libavdevice/oss.c | 4 +++-
libavdevice/oss.h | 1 +
libavdevice/oss_dec.c | 2 +-
3 files changed, 5 insertions(+), 2 deletions(-)
diff --git a/libavdevice/oss.c b/libavdevice/oss.c
index eddc2ddf1a..b042f58875 100644
--- a/libavdevice/oss.c
+++ b/libavdevice/oss.c
@@ -102,9 +102,11 @@ int ff_oss_audio_open(AVFormatContext *s1, int is_output,
switch(tmp) {
case AFMT_S16_LE:
s->codec_id = AV_CODEC_ID_PCM_S16LE;
+ s->sample_size = 2;
break;
case AFMT_S16_BE:
s->codec_id = AV_CODEC_ID_PCM_S16BE;
+ s->sample_size = 2;
break;
default:
av_log(s1, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
@@ -112,7 +114,7 @@ int ff_oss_audio_open(AVFormatContext *s1, int is_output,
return AVERROR(EIO);
}
err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
- CHECK_IOCTL_ERROR(SNDCTL_DSP_SETFMTS)
+ CHECK_IOCTL_ERROR(SNDCTL_DSP_SETFMT)
tmp = (s->channels == 2);
err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
diff --git a/libavdevice/oss.h b/libavdevice/oss.h
index 66d1a34cf6..f1da2b1bec 100644
--- a/libavdevice/oss.h
+++ b/libavdevice/oss.h
@@ -30,6 +30,7 @@ typedef struct OSSAudioData {
AVClass *class;
int fd;
int sample_rate;
+ int sample_size; /* in bytes ! */
int channels;
int frame_size; /* in bytes ! */
enum AVCodecID codec_id;
diff --git a/libavdevice/oss_dec.c b/libavdevice/oss_dec.c
index d3dbe77cf9..2cdc4324e8 100644
--- a/libavdevice/oss_dec.c
+++ b/libavdevice/oss_dec.c
@@ -91,7 +91,7 @@ static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
bdelay += abufi.bytes;
}
/* subtract time represented by the number of bytes in the audio fifo */
- cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
+ cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->sample_size * s->channels);
/* convert to wanted units */
pkt->pts = cur_time;
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