---
libavdevice/alsa.c | 2 +-
libavdevice/alsa_dec.c | 14 +++++++-------
libavdevice/alsa_enc.c | 10 +++++-----
3 files changed, 13 insertions(+), 13 deletions(-)
diff --git a/libavdevice/alsa.c b/libavdevice/alsa.c
index 6d68267..d394e43 100644
--- a/libavdevice/alsa.c
+++ b/libavdevice/alsa.c
@@ -194,7 +194,7 @@ av_cold int ff_alsa_open(AVFormatContext *ctx,
snd_pcm_stream_t mode,
snd_pcm_t *h;
snd_pcm_hw_params_t *hw_params;
snd_pcm_uframes_t buffer_size, period_size;
- uint64_t layout = ctx->streams[0]->codec->channel_layout;
+ uint64_t layout = ctx->streams[0]->codecpar->channel_layout;
if (ctx->filename[0] == 0) audio_device = "default";
else audio_device = ctx->filename;
diff --git a/libavdevice/alsa_dec.c b/libavdevice/alsa_dec.c
index e7819ec..58bf1dd 100644
--- a/libavdevice/alsa_dec.c
+++ b/libavdevice/alsa_dec.c
@@ -101,10 +101,10 @@ static av_cold int audio_read_header(AVFormatContext *s1)
}
/* take real parameters */
- st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
- st->codec->codec_id = codec_id;
- st->codec->sample_rate = s->sample_rate;
- st->codec->channels = s->channels;
+ st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
+ st->codecpar->codec_id = codec_id;
+ st->codecpar->sample_rate = s->sample_rate;
+ st->codecpar->channels = s->channels;
avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
return 0;
@@ -144,9 +144,9 @@ static int audio_read_packet(AVFormatContext *s1, AVPacket
*pkt)
snd_pcm_htimestamp(s->h, &ts_delay, ×tamp);
ts_delay += res;
pkt->pts = timestamp.tv_sec * 1000000LL
- + (timestamp.tv_nsec * st->codec->sample_rate
- - (int64_t)ts_delay * 1000000000LL + st->codec->sample_rate
* 500LL)
- / (st->codec->sample_rate * 1000LL);
+ + (timestamp.tv_nsec * st->codecpar->sample_rate
+ - (int64_t)ts_delay * 1000000000LL +
st->codecpar->sample_rate * 500LL)
+ / (st->codecpar->sample_rate * 1000LL);
pkt->size = res * s->frame_size;
diff --git a/libavdevice/alsa_enc.c b/libavdevice/alsa_enc.c
index 30c1719..3094b50 100644
--- a/libavdevice/alsa_enc.c
+++ b/libavdevice/alsa_enc.c
@@ -54,14 +54,14 @@ static av_cold int audio_write_header(AVFormatContext *s1)
int res;
st = s1->streams[0];
- sample_rate = st->codec->sample_rate;
- codec_id = st->codec->codec_id;
+ sample_rate = st->codecpar->sample_rate;
+ codec_id = st->codecpar->codec_id;
res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate,
- st->codec->channels, &codec_id);
- if (sample_rate != st->codec->sample_rate) {
+ st->codecpar->channels, &codec_id);
+ if (sample_rate != st->codecpar->sample_rate) {
av_log(s1, AV_LOG_ERROR,
"sample rate %d not available, nearest is %d\n",
- st->codec->sample_rate, sample_rate);
+ st->codecpar->sample_rate, sample_rate);
goto fail;
}
--
2.0.0
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