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| author | Vittorio Giovara <vittorio.giovara@gmail.com> | 2017-03-31 18:25:12 +0200 |
|---|---|---|
| committer | James Almer <jamrial@gmail.com> | 2022-03-15 09:42:36 -0300 |
| commit | 620d151e5cb095c3406a06a476d4b0bfaf7f0182 (patch) | |
| tree | 00dd1eca8d72cf227247ec0ca6a071c7157f3505 /libavformat/rtpenc.c | |
| parent | b76e878f5b64587afa584e16ed8353c26d9cf10f (diff) | |
| download | ffmpeg-620d151e5cb095c3406a06a476d4b0bfaf7f0182.tar.gz | |
rtp: convert to new channel layout API
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Diffstat (limited to 'libavformat/rtpenc.c')
| -rw-r--r-- | libavformat/rtpenc.c | 14 |
1 files changed, 7 insertions, 7 deletions
diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c index 6be67b5885..ce629a8095 100644 --- a/libavformat/rtpenc.c +++ b/libavformat/rtpenc.c @@ -236,7 +236,7 @@ static int rtp_write_header(AVFormatContext *s1) avpriv_set_pts_info(st, 32, 1, 8000); break; case AV_CODEC_ID_OPUS: - if (st->codecpar->channels > 2) { + if (st->codecpar->ch_layout.nb_channels > 2) { av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n"); goto fail; } @@ -264,7 +264,7 @@ static int rtp_write_header(AVFormatContext *s1) av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n"); goto fail; } - if (st->codecpar->channels != 1) { + if (st->codecpar->ch_layout.nb_channels != 1) { av_log(s1, AV_LOG_ERROR, "Only mono is supported\n"); goto fail; } @@ -541,24 +541,24 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) case AV_CODEC_ID_PCM_ALAW: case AV_CODEC_ID_PCM_U8: case AV_CODEC_ID_PCM_S8: - return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->channels); + return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->ch_layout.nb_channels); case AV_CODEC_ID_PCM_U16BE: case AV_CODEC_ID_PCM_U16LE: case AV_CODEC_ID_PCM_S16BE: case AV_CODEC_ID_PCM_S16LE: - return rtp_send_samples(s1, pkt->data, size, 16 * st->codecpar->channels); + return rtp_send_samples(s1, pkt->data, size, 16 * st->codecpar->ch_layout.nb_channels); case AV_CODEC_ID_PCM_S24BE: - return rtp_send_samples(s1, pkt->data, size, 24 * st->codecpar->channels); + return rtp_send_samples(s1, pkt->data, size, 24 * st->codecpar->ch_layout.nb_channels); case AV_CODEC_ID_ADPCM_G722: /* The actual sample size is half a byte per sample, but since the * stream clock rate is 8000 Hz while the sample rate is 16000 Hz, * the correct parameter for send_samples_bits is 8 bits per stream * clock. */ - return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->channels); + return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->ch_layout.nb_channels); case AV_CODEC_ID_ADPCM_G726: case AV_CODEC_ID_ADPCM_G726LE: return rtp_send_samples(s1, pkt->data, size, - st->codecpar->bits_per_coded_sample * st->codecpar->channels); + st->codecpar->bits_per_coded_sample * st->codecpar->ch_layout.nb_channels); case AV_CODEC_ID_MP2: case AV_CODEC_ID_MP3: rtp_send_mpegaudio(s1, pkt->data, size); |