mirror of https://git.videolan.org/git/ffmpeg.git
rtp: convert to new channel layout API
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com> Signed-off-by: James Almer <jamrial@gmail.com>
This commit is contained in:
parent
b76e878f5b
commit
620d151e5c
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@ -77,8 +77,11 @@ int ff_rtp_get_codec_info(AVCodecParameters *par, int payload_type)
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if (rtp_payload_types[i].codec_id != AV_CODEC_ID_NONE) {
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if (rtp_payload_types[i].codec_id != AV_CODEC_ID_NONE) {
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par->codec_type = rtp_payload_types[i].codec_type;
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par->codec_type = rtp_payload_types[i].codec_type;
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par->codec_id = rtp_payload_types[i].codec_id;
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par->codec_id = rtp_payload_types[i].codec_id;
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if (rtp_payload_types[i].audio_channels > 0)
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if (rtp_payload_types[i].audio_channels > 0) {
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par->channels = rtp_payload_types[i].audio_channels;
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av_channel_layout_uninit(&par->ch_layout);
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par->ch_layout.order = AV_CHANNEL_ORDER_UNSPEC;
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par->ch_layout.nb_channels = rtp_payload_types[i].audio_channels;
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}
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if (rtp_payload_types[i].clock_rate > 0)
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if (rtp_payload_types[i].clock_rate > 0)
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par->sample_rate = rtp_payload_types[i].clock_rate;
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par->sample_rate = rtp_payload_types[i].clock_rate;
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return 0;
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return 0;
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@ -111,13 +114,13 @@ int ff_rtp_get_payload_type(const AVFormatContext *fmt,
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/* G722 has 8000 as nominal rate even if the sample rate is 16000,
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/* G722 has 8000 as nominal rate even if the sample rate is 16000,
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* see section 4.5.2 in RFC 3551. */
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* see section 4.5.2 in RFC 3551. */
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if (par->codec_id == AV_CODEC_ID_ADPCM_G722 &&
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if (par->codec_id == AV_CODEC_ID_ADPCM_G722 &&
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par->sample_rate == 16000 && par->channels == 1)
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par->sample_rate == 16000 && par->ch_layout.nb_channels == 1)
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return rtp_payload_types[i].pt;
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return rtp_payload_types[i].pt;
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if (par->codec_type == AVMEDIA_TYPE_AUDIO &&
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if (par->codec_type == AVMEDIA_TYPE_AUDIO &&
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((rtp_payload_types[i].clock_rate > 0 &&
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((rtp_payload_types[i].clock_rate > 0 &&
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par->sample_rate != rtp_payload_types[i].clock_rate) ||
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par->sample_rate != rtp_payload_types[i].clock_rate) ||
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(rtp_payload_types[i].audio_channels > 0 &&
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(rtp_payload_types[i].audio_channels > 0 &&
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par->channels != rtp_payload_types[i].audio_channels)))
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par->ch_layout.nb_channels != rtp_payload_types[i].audio_channels)))
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continue;
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continue;
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return rtp_payload_types[i].pt;
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return rtp_payload_types[i].pt;
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}
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}
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@ -538,7 +538,7 @@ static int opus_write_extradata(AVCodecParameters *codecpar)
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* This mapping family only supports mono and stereo layouts. And RFC7587
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* This mapping family only supports mono and stereo layouts. And RFC7587
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* specifies that the number of channels in the SDP must be 2.
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* specifies that the number of channels in the SDP must be 2.
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*/
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*/
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if (codecpar->channels > 2) {
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if (codecpar->ch_layout.nb_channels > 2) {
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return AVERROR_INVALIDDATA;
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return AVERROR_INVALIDDATA;
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}
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}
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@ -553,7 +553,7 @@ static int opus_write_extradata(AVCodecParameters *codecpar)
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/* Version */
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/* Version */
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bytestream_put_byte (&bs, 0x1);
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bytestream_put_byte (&bs, 0x1);
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/* Channel count */
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/* Channel count */
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bytestream_put_byte (&bs, codecpar->channels);
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bytestream_put_byte (&bs, codecpar->ch_layout.nb_channels);
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/* Pre skip */
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/* Pre skip */
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bytestream_put_le16 (&bs, 0);
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bytestream_put_le16 (&bs, 0);
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/* Input sample rate */
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/* Input sample rate */
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@ -64,11 +64,11 @@ static int amr_handle_packet(AVFormatContext *ctx, PayloadContext *data,
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return AVERROR_INVALIDDATA;
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return AVERROR_INVALIDDATA;
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}
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}
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if (st->codecpar->channels != 1) {
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if (st->codecpar->ch_layout.nb_channels != 1) {
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av_log(ctx, AV_LOG_ERROR, "Only mono AMR is supported\n");
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av_log(ctx, AV_LOG_ERROR, "Only mono AMR is supported\n");
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return AVERROR_INVALIDDATA;
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return AVERROR_INVALIDDATA;
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}
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}
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st->codecpar->channel_layout = AV_CH_LAYOUT_MONO;
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av_channel_layout_default(&st->codecpar->ch_layout, 1);
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/* The AMR RTP packet consists of one header byte, followed
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/* The AMR RTP packet consists of one header byte, followed
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* by one TOC byte for each AMR frame in the packet, followed
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* by one TOC byte for each AMR frame in the packet, followed
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@ -236,7 +236,7 @@ static int rtp_write_header(AVFormatContext *s1)
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avpriv_set_pts_info(st, 32, 1, 8000);
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avpriv_set_pts_info(st, 32, 1, 8000);
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break;
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break;
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case AV_CODEC_ID_OPUS:
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case AV_CODEC_ID_OPUS:
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if (st->codecpar->channels > 2) {
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if (st->codecpar->ch_layout.nb_channels > 2) {
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av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
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av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
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goto fail;
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goto fail;
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}
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}
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@ -264,7 +264,7 @@ static int rtp_write_header(AVFormatContext *s1)
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av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
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av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
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goto fail;
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goto fail;
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}
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}
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if (st->codecpar->channels != 1) {
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if (st->codecpar->ch_layout.nb_channels != 1) {
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av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
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av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
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goto fail;
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goto fail;
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}
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}
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@ -541,24 +541,24 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
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case AV_CODEC_ID_PCM_ALAW:
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case AV_CODEC_ID_PCM_ALAW:
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case AV_CODEC_ID_PCM_U8:
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case AV_CODEC_ID_PCM_U8:
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case AV_CODEC_ID_PCM_S8:
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case AV_CODEC_ID_PCM_S8:
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return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->channels);
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return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->ch_layout.nb_channels);
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case AV_CODEC_ID_PCM_U16BE:
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case AV_CODEC_ID_PCM_U16BE:
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case AV_CODEC_ID_PCM_U16LE:
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case AV_CODEC_ID_PCM_U16LE:
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case AV_CODEC_ID_PCM_S16BE:
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case AV_CODEC_ID_PCM_S16BE:
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case AV_CODEC_ID_PCM_S16LE:
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case AV_CODEC_ID_PCM_S16LE:
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return rtp_send_samples(s1, pkt->data, size, 16 * st->codecpar->channels);
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return rtp_send_samples(s1, pkt->data, size, 16 * st->codecpar->ch_layout.nb_channels);
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case AV_CODEC_ID_PCM_S24BE:
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case AV_CODEC_ID_PCM_S24BE:
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return rtp_send_samples(s1, pkt->data, size, 24 * st->codecpar->channels);
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return rtp_send_samples(s1, pkt->data, size, 24 * st->codecpar->ch_layout.nb_channels);
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case AV_CODEC_ID_ADPCM_G722:
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case AV_CODEC_ID_ADPCM_G722:
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/* The actual sample size is half a byte per sample, but since the
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/* The actual sample size is half a byte per sample, but since the
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* stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
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* stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
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* the correct parameter for send_samples_bits is 8 bits per stream
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* the correct parameter for send_samples_bits is 8 bits per stream
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* clock. */
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* clock. */
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return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->channels);
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return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->ch_layout.nb_channels);
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case AV_CODEC_ID_ADPCM_G726:
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case AV_CODEC_ID_ADPCM_G726:
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case AV_CODEC_ID_ADPCM_G726LE:
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case AV_CODEC_ID_ADPCM_G726LE:
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return rtp_send_samples(s1, pkt->data, size,
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return rtp_send_samples(s1, pkt->data, size,
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st->codecpar->bits_per_coded_sample * st->codecpar->channels);
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st->codecpar->bits_per_coded_sample * st->codecpar->ch_layout.nb_channels);
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case AV_CODEC_ID_MP2:
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case AV_CODEC_ID_MP2:
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case AV_CODEC_ID_MP3:
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case AV_CODEC_ID_MP3:
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rtp_send_mpegaudio(s1, pkt->data, size);
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rtp_send_mpegaudio(s1, pkt->data, size);
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@ -323,19 +323,19 @@ static int sdp_parse_rtpmap(AVFormatContext *s,
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case AVMEDIA_TYPE_AUDIO:
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case AVMEDIA_TYPE_AUDIO:
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av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
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av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
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par->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
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par->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
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par->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
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par->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO;
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if (i > 0) {
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if (i > 0) {
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par->sample_rate = i;
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par->sample_rate = i;
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avpriv_set_pts_info(st, 32, 1, par->sample_rate);
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avpriv_set_pts_info(st, 32, 1, par->sample_rate);
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get_word_sep(buf, sizeof(buf), "/", &p);
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get_word_sep(buf, sizeof(buf), "/", &p);
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i = atoi(buf);
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i = atoi(buf);
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if (i > 0)
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if (i > 0)
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par->channels = i;
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av_channel_layout_default(&par->ch_layout, i);
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}
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}
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av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
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av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
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par->sample_rate);
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par->sample_rate);
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av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
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av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
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par->channels);
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par->ch_layout.nb_channels);
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break;
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break;
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case AVMEDIA_TYPE_VIDEO:
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case AVMEDIA_TYPE_VIDEO:
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av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
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av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
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@ -75,7 +75,6 @@ enum RTSPControlTransport {
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#define RTSPS_DEFAULT_PORT 322
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#define RTSPS_DEFAULT_PORT 322
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#define RTSP_MAX_TRANSPORTS 8
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#define RTSP_MAX_TRANSPORTS 8
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#define RTSP_TCP_MAX_PACKET_SIZE 1472
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#define RTSP_TCP_MAX_PACKET_SIZE 1472
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#define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1
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#define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
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#define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
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#define RTSP_RTP_PORT_MIN 5000
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#define RTSP_RTP_PORT_MIN 5000
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#define RTSP_RTP_PORT_MAX 65000
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#define RTSP_RTP_PORT_MAX 65000
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