diff --git a/trunk/conf/rtc.tcp.only.conf b/trunk/conf/rtc.tcp.only.conf
new file mode 100644
index 0000000000..a18474c8cc
--- /dev/null
+++ b/trunk/conf/rtc.tcp.only.conf
@@ -0,0 +1,46 @@
+# WebRTC streaming config for SRS.
+# @see full.conf for detail config.
+
+listen 1935;
+max_connections 1000;
+daemon off;
+srs_log_tank console;
+
+http_server {
+ enabled on;
+ listen 8080;
+ dir ./objs/nginx/html;
+}
+
+http_api {
+ enabled on;
+ listen 1985;
+}
+stats {
+ network 0;
+}
+rtc_server {
+ enabled on;
+ tcp {
+ enabled on;
+ listen 8000;
+ }
+ protocol tcp;
+ # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
+ candidate $CANDIDATE;
+}
+
+vhost __defaultVhost__ {
+ rtc {
+ enabled on;
+ # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtmp-to-rtc
+ rtmp_to_rtc off;
+ # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtc-to-rtmp
+ rtc_to_rtmp off;
+ }
+ http_remux {
+ enabled on;
+ mount [vhost]/[app]/[stream].flv;
+ }
+}
+
diff --git a/trunk/conf/rtc.tcp.udp.conf b/trunk/conf/rtc.tcp.udp.conf
new file mode 100644
index 0000000000..7fcd3a808a
--- /dev/null
+++ b/trunk/conf/rtc.tcp.udp.conf
@@ -0,0 +1,47 @@
+# WebRTC streaming config for SRS.
+# @see full.conf for detail config.
+
+listen 1935;
+max_connections 1000;
+daemon off;
+srs_log_tank console;
+
+http_server {
+ enabled on;
+ listen 8080;
+ dir ./objs/nginx/html;
+}
+
+http_api {
+ enabled on;
+ listen 1985;
+}
+stats {
+ network 0;
+}
+rtc_server {
+ enabled on;
+ listen 8000; # UDP port
+ tcp {
+ enabled on;
+ listen 8000;
+ }
+ protocol all;
+ # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
+ candidate $CANDIDATE;
+}
+
+vhost __defaultVhost__ {
+ rtc {
+ enabled on;
+ # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtmp-to-rtc
+ rtmp_to_rtc off;
+ # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtc-to-rtmp
+ rtc_to_rtmp off;
+ }
+ http_remux {
+ enabled on;
+ mount [vhost]/[app]/[stream].flv;
+ }
+}
+
diff --git a/trunk/doc/CHANGELOG.md b/trunk/doc/CHANGELOG.md
index 52b6db046a..676bd5f1d3 100644
--- a/trunk/doc/CHANGELOG.md
+++ b/trunk/doc/CHANGELOG.md
@@ -7,6 +7,7 @@ The changelog for SRS.
## SRS 5.0 Changelog
+* v5.0, 2023-10-25, Merge [#3845](https://github.com/ossrs/srs/pull/3845): RTC: Fix FFmpeg opus audio noisy issue. v5.0.195 (#3845)
* v5.0, 2023-10-21, Merge [#3847](https://github.com/ossrs/srs/pull/3847): WebRTC: TCP transport should use read_fully instead of read. v5.0.194 (#3847)
* v5.0, 2023-10-20, Merge [#3846](https://github.com/ossrs/srs/pull/3846): Added system library option for ffmpeg, srtp, srt libraries. v5.0.193 (#3846)
* v5.0, 2023-10-17, Merge [#3840](https://github.com/ossrs/srs/pull/3840): Disable asan by default. v5.0.192 (#3840)
diff --git a/trunk/ide/srs_clion/CMakeLists.txt b/trunk/ide/srs_clion/CMakeLists.txt
index b0a9249e3e..a4d4243955 100755
--- a/trunk/ide/srs_clion/CMakeLists.txt
+++ b/trunk/ide/srs_clion/CMakeLists.txt
@@ -44,7 +44,6 @@ set(DEPS_LIBS ${SRS_DIR}/objs/st/libst.a
${SRS_DIR}/objs/srtp2/lib/libsrtp2.a
${SRS_DIR}/objs/ffmpeg/lib/libavcodec.a
${SRS_DIR}/objs/ffmpeg/lib/libavutil.a
- ${SRS_DIR}/objs/opus/lib/libopus.a
${SRS_DIR}/objs/ffmpeg/lib/libswresample.a
${SRS_DIR}/objs/srt/lib/libsrt.a)
foreach(DEPS_LIB ${DEPS_LIBS})
diff --git a/trunk/src/app/srs_app_rtc_codec.cpp b/trunk/src/app/srs_app_rtc_codec.cpp
index 840ace5253..cbc46d1388 100644
--- a/trunk/src/app/srs_app_rtc_codec.cpp
+++ b/trunk/src/app/srs_app_rtc_codec.cpp
@@ -242,7 +242,7 @@ srs_error_t SrsAudioTranscoder::init_enc(SrsAudioCodecId dst_codec, int dst_chan
enc_->channel_layout = av_get_default_channel_layout(dst_channels);
enc_->bit_rate = dst_bit_rate;
enc_->sample_fmt = codec->sample_fmts[0];
- enc_->time_base.num = 1; enc_->time_base.den = 1000; // {1, 1000}
+ enc_->time_base.num = 1; enc_->time_base.den = dst_samplerate; // {1, dst_samplerate}
if (dst_codec == SrsAudioCodecIdOpus) {
//TODO: for more level setting
enc_->compression_level = 1;
@@ -261,14 +261,6 @@ srs_error_t SrsAudioTranscoder::init_enc(SrsAudioCodecId dst_codec, int dst_chan
return srs_error_new(ERROR_RTC_RTP_MUXER, "Could not allocate audio encode in frame");
}
- enc_frame_->format = enc_->sample_fmt;
- enc_frame_->nb_samples = enc_->frame_size;
- enc_frame_->channel_layout = enc_->channel_layout;
-
- if (av_frame_get_buffer(enc_frame_, 0) < 0) {
- return srs_error_new(ERROR_RTC_RTP_MUXER, "Could not get audio frame buffer");
- }
-
enc_packet_ = av_packet_alloc();
if (!enc_packet_) {
return srs_error_new(ERROR_RTC_RTP_MUXER, "Could not allocate audio encode out packet");
@@ -380,25 +372,35 @@ srs_error_t SrsAudioTranscoder::encode(std::vector &pkts)
if (next_out_pts_ == AV_NOPTS_VALUE) {
next_out_pts_ = new_pkt_pts_;
} else {
- int64_t diff = llabs(new_pkt_pts_ - next_out_pts_);
+ int64_t diff = llabs(new_pkt_pts_ - av_rescale(next_out_pts_, 1000, enc_->time_base.den));
if (diff > 1000) {
srs_trace("time diff to large=%lld, next out=%lld, new pkt=%lld, set to new pkt",
diff, next_out_pts_, new_pkt_pts_);
- next_out_pts_ = new_pkt_pts_;
+ next_out_pts_ = av_rescale(new_pkt_pts_, enc_->time_base.den, 1000);
}
}
- int frame_cnt = 0;
while (av_audio_fifo_size(fifo_) >= enc_->frame_size) {
+ enc_frame_->format = enc_->sample_fmt;
+ enc_frame_->nb_samples = enc_->frame_size;
+ enc_frame_->channel_layout = enc_->channel_layout;
+
+ if (av_frame_get_buffer(enc_frame_, 0) < 0) {
+ av_frame_free(&enc_frame_);
+ return srs_error_new(ERROR_RTC_RTP_MUXER, "Could not get audio frame buffer");
+ }
+
/* Read as many samples from the FIFO buffer as required to fill the frame.
* The samples are stored in the frame temporarily. */
if (av_audio_fifo_read(fifo_, (void **)enc_frame_->data, enc_->frame_size) < enc_->frame_size) {
+ av_frame_free(&enc_frame_);
return srs_error_new(ERROR_RTC_RTP_MUXER, "Could not read data from FIFO");
}
/* send the frame for encoding */
- enc_frame_->pts = next_out_pts_ + av_rescale(enc_->frame_size * frame_cnt, 1000, enc_->sample_rate);
- ++frame_cnt;
+ enc_frame_->pts = next_out_pts_;
+ next_out_pts_ += enc_->frame_size;
int error = avcodec_send_frame(enc_, enc_frame_);
+ av_frame_unref(enc_frame_);
if (error < 0) {
return srs_error_new(ERROR_RTC_RTP_MUXER, "Error sending the frame to the encoder(%d,%s)", error,
av_make_error_string(err_buf, AV_ERROR_MAX_STRING_SIZE, error));
@@ -419,6 +421,10 @@ srs_error_t SrsAudioTranscoder::encode(std::vector &pkts)
av_make_error_string(err_buf, AV_ERROR_MAX_STRING_SIZE, error));
}
+ // rescale time base from sample_rate 1000.
+ enc_packet_->dts = av_rescale(enc_packet_->dts, 1000, enc_->time_base.den);
+ enc_packet_->pts = av_rescale(enc_packet_->pts, 1000, enc_->time_base.den);
+
SrsAudioFrame *out_frame = new SrsAudioFrame;
char *buf = new char[enc_packet_->size];
memcpy(buf, enc_packet_->data, enc_packet_->size);
@@ -429,8 +435,6 @@ srs_error_t SrsAudioTranscoder::encode(std::vector &pkts)
}
}
- next_out_pts_ += av_rescale(enc_->frame_size * frame_cnt, 1000, enc_->sample_rate);
-
return srs_success;
}
diff --git a/trunk/src/app/srs_app_source.cpp b/trunk/src/app/srs_app_source.cpp
index a7f45bff5d..321ce41cd4 100755
--- a/trunk/src/app/srs_app_source.cpp
+++ b/trunk/src/app/srs_app_source.cpp
@@ -2733,7 +2733,7 @@ srs_error_t SrsLiveSource::consumer_dumps(SrsLiveConsumer* consumer, bool ds, bo
// print status.
if (dg) {
- srs_trace("create consumer, active=%d, queue_size=%.2f, jitter=%d", hub->active(), queue_size, jitter_algorithm);
+ srs_trace("create consumer, active=%d, queue_size=%dms, jitter=%d", hub->active(), srsu2msi(queue_size), jitter_algorithm);
} else {
srs_trace("create consumer, active=%d, ignore gop cache, jitter=%d", hub->active(), jitter_algorithm);
}
diff --git a/trunk/src/core/srs_core_version5.hpp b/trunk/src/core/srs_core_version5.hpp
index 663dd68105..83588407d8 100644
--- a/trunk/src/core/srs_core_version5.hpp
+++ b/trunk/src/core/srs_core_version5.hpp
@@ -9,6 +9,6 @@
#define VERSION_MAJOR 5
#define VERSION_MINOR 0
-#define VERSION_REVISION 194
+#define VERSION_REVISION 195
#endif
diff --git a/trunk/src/protocol/srs_protocol_rtmp_handshake.cpp b/trunk/src/protocol/srs_protocol_rtmp_handshake.cpp
index f904555814..0754d7724e 100644
--- a/trunk/src/protocol/srs_protocol_rtmp_handshake.cpp
+++ b/trunk/src/protocol/srs_protocol_rtmp_handshake.cpp
@@ -298,8 +298,6 @@ namespace srs_internal
{
srs_error_t err = srs_success;
- int32_t bits_count = 1024;
-
close();
//1. Create the DH