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RTC: Fix FFmpeg opus audio noisy issue. v5.0.195 v6.0.95 #3845

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merged 8 commits into from
Oct 25, 2023
Merged

RTC: Fix FFmpeg opus audio noisy issue. v5.0.195 v6.0.95 #3845

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00027b6
RTC: Fix FFmpeg opus audio noisy issue
xiaozhihong Oct 19, 2023
758a8f6
Enable --ffmpeg-opus=on by default.
winlinvip Oct 21, 2023
8ad925c
Fix warning.
winlinvip Oct 21, 2023
67f5cf4
Fix log bug.
winlinvip Oct 21, 2023
0c568a7
Optimize the code.
duiniuluantanqin Oct 22, 2023
0765532
Add conf for rtc.
winlinvip Oct 25, 2023
285260b
Merge branch 'develop' into pr-bugfix/ffmpeg_opus
winlinvip Oct 25, 2023
9e33fb6
Update release to v5.0.195 v6.0.95
winlinvip Oct 25, 2023
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Optimize the code.
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@duiniuluantanqin
duiniuluantanqin committed Oct 22, 2023
commit 0c568a70d7c439bc2809bb53c3533279d004abd2
1 change: 0 additions & 1 deletion trunk/ide/srs_clion/CMakeLists.txt
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Original file line number Diff line number Diff line change
Expand Up @@ -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})
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37 changes: 21 additions & 16 deletions trunk/src/app/srs_app_rtc_codec.cpp
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Original file line number Diff line number Diff line change
Expand Up @@ -90,6 +90,7 @@ SrsAudioTranscoder::SrsAudioTranscoder()
dec_frame_ = NULL;
dec_packet_ = NULL;
enc_ = NULL;
enc_frame_ = NULL;
enc_packet_ = NULL;
swr_ = NULL;
swr_data_ = NULL;
Expand Down Expand Up @@ -122,6 +123,10 @@ SrsAudioTranscoder::~SrsAudioTranscoder()
avcodec_free_context(&enc_);
}

if (enc_frame_) {
av_frame_free(&enc_frame_);
}

if (enc_packet_) {
av_packet_free(&enc_packet_);
}
Expand Down Expand Up @@ -251,6 +256,11 @@ srs_error_t SrsAudioTranscoder::init_enc(SrsAudioCodecId dst_codec, int dst_chan
return srs_error_new(ERROR_RTC_RTP_MUXER, "Could not open codec");
}

enc_frame_ = av_frame_alloc();
if (!enc_frame_) {
return srs_error_new(ERROR_RTC_RTP_MUXER, "Could not allocate audio encode in frame");
}

enc_packet_ = av_packet_alloc();
if (!enc_packet_) {
return srs_error_new(ERROR_RTC_RTP_MUXER, "Could not allocate audio encode out packet");
Expand Down Expand Up @@ -371,31 +381,26 @@ srs_error_t SrsAudioTranscoder::encode(std::vector<SrsAudioFrame*> &pkts)
}

while (av_audio_fifo_size(fifo_) >= enc_->frame_size) {
AVFrame* enc_frame = av_frame_alloc();
if (!enc_frame) {
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;
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);
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);
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_;
enc_frame_->pts = next_out_pts_;
next_out_pts_ += enc_->frame_size;
int error = avcodec_send_frame(enc_, enc_frame);
av_frame_free(&enc_frame);
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));
Expand All @@ -418,7 +423,7 @@ srs_error_t SrsAudioTranscoder::encode(std::vector<SrsAudioFrame*> &pkts)

// 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);
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];
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1 change: 1 addition & 0 deletions trunk/src/app/srs_app_rtc_codec.hpp
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Expand Up @@ -38,6 +38,7 @@ class SrsAudioTranscoder
AVPacket *dec_packet_;

AVCodecContext *enc_;
AVFrame *enc_frame_;
AVPacket *enc_packet_;

SwrContext *swr_;
Expand Down
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