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Code Issues Releases Activity

avdevice/oss_dec: account for sample size when computing timestamp

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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@passwd.hu>
This commit is contained in:
Matt Jacobson 2022-06-01 05:06:16 -04:00 committed by Marton Balint
parent fee765c207
commit b3e261bab3
3 changed files with 5 additions and 2 deletions
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4
libavdevice/oss.c
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@ -102,9 +102,11 @@ int ff_oss_audio_open(AVFormatContext *s1, int is_output,
switch(tmp) { switch(tmp) {
case AFMT_S16_LE: case AFMT_S16_LE:
s->codec_id = AV_CODEC_ID_PCM_S16LE; s->codec_id = AV_CODEC_ID_PCM_S16LE;
s->sample_size = 2;
break; break;
case AFMT_S16_BE: case AFMT_S16_BE:
s->codec_id = AV_CODEC_ID_PCM_S16BE; s->codec_id = AV_CODEC_ID_PCM_S16BE;
s->sample_size = 2;
break; break;
default: default:
av_log(s1, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n"); 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); return AVERROR(EIO);
} }
err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp); err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
CHECK_IOCTL_ERROR(SNDCTL_DSP_SETFMTS) CHECK_IOCTL_ERROR(SNDCTL_DSP_SETFMT)
tmp = (s->channels == 2); tmp = (s->channels == 2);
err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp); err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);

1
libavdevice/oss.h
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@ -30,6 +30,7 @@ typedef struct OSSAudioData {
AVClass *class; AVClass *class;
int fd; int fd;
int sample_rate; int sample_rate;
int sample_size; /* in bytes ! */
int channels; int channels;
int frame_size; /* in bytes ! */ int frame_size; /* in bytes ! */
enum AVCodecID codec_id; enum AVCodecID codec_id;

2
libavdevice/oss_dec.c
View File

@ -91,7 +91,7 @@ static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
bdelay += abufi.bytes; bdelay += abufi.bytes;
} }
/* subtract time represented by the number of bytes in the audio fifo */ /* 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 */ /* convert to wanted units */
pkt->pts = cur_time; pkt->pts = cur_time;