diff --git a/src/audio/codec.cpp b/src/audio/codec.cpp index 0f2989b5d..3ba922150 100644 --- a/src/audio/codec.cpp +++ b/src/audio/codec.cpp @@ -3,7 +3,7 @@ * @author Martin Pulec */ /* - * Copyright (c) 2013-2021 CESNET, z. s. p. o. + * Copyright (c) 2013-2023 CESNET, z. s. p. o. * All rights reserved. * * Redistribution and use in source and binary forms, with or without @@ -65,7 +65,7 @@ static const unordered_map> audio_c {AC_ALAW, { "A-law", 0x0006 }}, {AC_MULAW, { "u-law", 0x0007 }}, {AC_SPEEX, { "speex", 0xA109 }}, - {AC_OPUS, { "OPUS", 0x7375704F }}, // == Opus, the TwoCC isn't defined + {AC_OPUS, { "Opus", 0x7375704F }}, // == Opus, the TwoCC isn't defined {AC_G722, { "G.722", 0x028F }}, {AC_MP3, { "MP3", 0x0055 }}, {AC_AAC, { "AAC", 0x00FF }}, diff --git a/src/audio/codec/libavcodec.c b/src/audio/codec/libavcodec.c index 2be0d5abc..34485e330 100644 --- a/src/audio/codec/libavcodec.c +++ b/src/audio/codec/libavcodec.c @@ -1,5 +1,5 @@ /** - * @file audio/codec/libavcodec.cpp + * @file audio/codec/libavcodec.c * @author Martin Pulec */ /* @@ -335,7 +335,7 @@ static bool reinitialize_encoder(struct libavcodec_codec_state *s, struct audio_ if (strcmp(s->codec->name, "libopus") == 0) { int ret = av_opt_set(s->codec_ctx->priv_data, "application", "lowdelay", 0); if (ret != 0) { - print_libav_audio_error(LOG_LEVEL_WARNING, "Could not set OPUS low delay app type", ret); + print_libav_audio_error(LOG_LEVEL_WARNING, "Could not set Opus low delay app type", ret); } } else if (strcmp(s->codec->name, "opus") == 0) { char warn[] = MOD_NAME "Native FFmpeg Opus encoder seems to be currently broken " @@ -356,7 +356,7 @@ static bool reinitialize_encoder(struct libavcodec_codec_state *s, struct audio_ if (s->codec->id == AV_CODEC_ID_OPUS) { int ret = av_opt_set_double(s->codec_ctx->priv_data, "frame_duration", frame_duration, 0); if (ret != 0) { - print_libav_audio_error(LOG_LEVEL_ERROR, "Could not set OPUS frame duration", ret); + print_libav_audio_error(LOG_LEVEL_ERROR, "Could not set Opus frame duration", ret); } } if (s->codec->id == AV_CODEC_ID_FLAC) { diff --git a/src/main.cpp b/src/main.cpp index 75d215f6f..1f7414ff5 100644 --- a/src/main.cpp +++ b/src/main.cpp @@ -1130,7 +1130,7 @@ static int adjust_params(struct ug_options *opt) { if (opt->audio.codec_cfg == nullptr) { if (strcasecmp(opt->audio.proto, "rtsp") == 0 || strcasecmp(opt->audio.proto, "sdp") == 0) { - opt->audio.codec_cfg = "OPUS:sample_rate=48000"; + opt->audio.codec_cfg = "Opus:sample_rate=48000"; } else { opt->audio.codec_cfg = DEFAULT_AUDIO_CODEC; } diff --git a/src/rtsp/BasicRTSPOnlySubsession.cpp b/src/rtsp/BasicRTSPOnlySubsession.cpp index bed3f49cb..eddf38f18 100644 --- a/src/rtsp/BasicRTSPOnlySubsession.cpp +++ b/src/rtsp/BasicRTSPOnlySubsession.cpp @@ -170,7 +170,7 @@ void BasicRTSPOnlySubsession::setSDPLines() { //rtpmapLine, // a=rtpmap:... (if present) rtp_port_audio + 1, rtpPayloadType, - audio_codec == AC_MULAW ? "PCMU" : audio_codec == AC_ALAW ? "PCMA" : "OPUS", + audio_codec == AC_MULAW ? "PCMU" : audio_codec == AC_ALAW ? "PCMA" : "opus", audio_sample_rate, audio_channels, trackId()); // a=control: diff --git a/src/transmit.cpp b/src/transmit.cpp index 6ff990229..d57b31dd8 100644 --- a/src/transmit.cpp +++ b/src/transmit.cpp @@ -17,7 +17,7 @@ * * Copyright (c) 2005-2010 Fundació i2CAT, Internet I Innovació Digital a Catalunya * Copyright (c) 2001-2004 University of Southern California - * Copyright (c) 2005-2021 CESNET z.s.p.o. + * Copyright (c) 2005-2023 CESNET z.s.p.o. * * Redistribution and use in source and binary forms, with or without * modification, is permitted provided that the following conditions @@ -983,7 +983,7 @@ void audio_tx_send_standard(struct tx* tx, struct rtp *rtp_session, if (buffer->get_codec() == AC_OPUS) { // OPUS needs to fit one package if (payload_size < data_len) { - log_msg(LOG_LEVEL_ERROR, "Transmit: OPUS frame larger than packet! Discarding...\n"); + log_msg(LOG_LEVEL_ERROR, "Transmit: Opus frame larger than packet! Discarding...\n"); return; } } else { // we may split the data into more packets, compute chunk size @@ -997,8 +997,8 @@ void audio_tx_send_standard(struct tx* tx, struct rtp *rtp_session, // interleave if (buffer->get_codec() == AC_OPUS) { - if (buffer->get_channel_count() > 1) { // we cannot interleave OPUS here - LOG(LOG_LEVEL_ERROR) << "Transmit: Only OPUS with 1 channel is supported in RFC-compliant mode! Discarding...\n"; + if (buffer->get_channel_count() > 1) { // we cannot interleave Opus here + LOG(LOG_LEVEL_ERROR) << "Transmit: Only Opus with 1 channel is supported in RFC-compliant mode! Discarding...\n"; return; } memcpy(tx->tmp_packet, buffer->get_data(0), pkt_len); diff --git a/src/utils/sdp.c b/src/utils/sdp.c index 1aeb668b3..ecbb5d58f 100644 --- a/src/utils/sdp.c +++ b/src/utils/sdp.c @@ -183,11 +183,11 @@ int sdp_add_audio(struct sdp *sdp, int port, int sample_rate, int channels, audi audio_codec = "PCMU"; break; case AC_OPUS: - audio_codec = "OPUS"; - ts_rate = 48000; // RFC 7587 specifies always 48 kHz for OPUS + audio_codec = "opus"; + ts_rate = 48000; // RFC 7587 specifies always 48 kHz for Opus break; default: - log_msg(LOG_LEVEL_ERROR, "[SDP] Currently only PCMA, PCMU and OPUS audio codecs are supported!\n"); + log_msg(LOG_LEVEL_ERROR, "[SDP] Currently only PCMA, PCMU and Opus audio codecs are supported!\n"); return -2; }