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

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Follow the example in FFmpeg's doc, before calling the API
`avcodec_send_frame`, always use `av_frame_alloc` to create a new frame.

---------

Co-authored-by: Haibo Chen <495810242@qq.com>
This commit is contained in:
john 2023-10-25 11:55:32 +08:00 • committed by GitHub
parent 9b07d840ed
commit 9238f09b0b
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GPG key ID: 4AEE18F83AFDEB23
10 changed files with 119 additions and 23 deletions
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36
trunk/src/app/srs_app_rtc_codec.cpp
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@ -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<SrsAudioFrame*> &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<SrsAudioFrame*> &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<SrsAudioFrame*> &pkts)
}
}
next_out_pts_ += av_rescale(enc_->frame_size * frame_cnt, 1000, enc_->sample_rate);
return srs_success;
}