PR: RTMP push streaming, WebRTC playback, dual-channel audio has issues, single-channel audio is normal.

[18981707877]

Description'

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Please describe the issue you encountered here.
'
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1. SRS version: 4.0.62
2. The log of SRS is as follows:

The log is normal.

1. The configuration of SRS is as follows (Config):

daemon              off;
listen              1935;
max_connections     1000;
srs_log_file        srs.log;

http_server {
    enabled         on;
    listen          8088;
    dir             html;
}

http_api {
    enabled         on;
    listen          1985;
}
stats {
    network         0;
}

# RTSP
stream_caster {
    # whether stream caster is enabled.
    # default: off
    enabled         on;
    # the caster type of stream, the casters:
    #       rtsp, Real Time Streaming Protocol (RTSP).
    caster          rtsp;
    # the output rtmp url.
    # for rtsp caster, the typically output url:
    #           rtmp://127.0.0.1/[app]/[stream]
    #       for example, the rtsp url:
    #           rtsp://192.168.1.173:8544/live/livestream.sdp
    #       where the [app] is "live" and [stream] is "livestream", output is:
    #           rtmp://127.0.0.1/live/livestream
    output          rtmp://127.0.0.1:1936/[app]/[stream];
    # the listen port for stream caster.
    #       for rtsp caster, listen at tcp port. for example, 554.
    listen          1554;
    # for the rtsp caster, the rtp server local port over udp,
    # which reply the rtsp setup request message, the port will be used:
    #       [rtp_port_min, rtp_port_max)
    rtp_port_min    57200;
    rtp_port_max    57300;
}

rtc_server {
    enabled         on;
    # Listen at udp://8000
    listen          8000;
    #
    # The $CANDIDATE means fetch from env, if not configed, use * as default.
    #
    # The * means retrieving server IP automatically, from all network interfaces,
    # @see https://github.com/ossrs/srs/issues/307#issuecomment-599028124
    candidate       $CANDIDATE;
}

vhost __defaultVhost__ {
    gop_cache       off;
    queue_length    10;
    min_latency     on;
    mr {
        enabled     off;
    }
    mw_latency      100;
    tcp_nodelay     on;
    rtc {
        enabled     on;
        bframe      discard;
    }
    http_remux {
        enabled     on;
        mount       [vhost]/[app]/[stream].flv;
        hstrs       on;
    }
}

Replay

How to replay bug?

Steps to reproduce the bug

1. ffmpeg -re -i D:\code\ffmpegabout\ffmpeg4.2\ffmpeg-20200223-90913ab-win32-static\bin\123.mp4 -vcodec libx264 -bf 0 -x264-params keyint=120 -acodec aac -ac 2 -ar 48000 -f flv rtmp://192.168.1.165:1935/live/669E0DB78AC84B86B031E587FB330C2B
2. 1. ffmpeg -re -i D:\code\ffmpegabout\ffmpeg4.2\ffmpeg-20200223-90913ab-win32-static\bin\123.mp4 -vcodec libx264 -bf 0 -x264-params keyint=120 -acodec aac -ac 1 -ar 48000 -f flv rtmp://192.168.1.165:1935/live/669E0DB78AC84B86B031E587FB330C2B
      Audio playback is normal when using 2 WebRTC streams for pushing, but there is noise when using 1 WebRTC stream for pushing.

The repair process is as follows: it is necessary to determine whether the decoded audio data is in plane format. Based on this situation, copy the left and right channel data to contiguous memory space. When resampling, the left and right channel data should also be organized according to this rule. The processing process of the completed PCM data is the same as above.

git diff 4bb3ad5637b5541df515fa9a9d69143310397c60
diff --git a/trunk/src/app/srs_app_rtc_codec.cpp b/trunk/src/app/srs_app_rtc_codec.cpp
index b3db983..ceb3837 100644
--- a/trunk/src/app/srs_app_rtc_codec.cpp
+++ b/trunk/src/app/srs_app_rtc_codec.cpp
@@ -129,13 +129,24 @@ srs_error_t SrsAudioDecoder::decode(SrsSample *pkt, char *buf, int &size)
         if (pcm_size < 0) {
             return srs_error_new(ERROR_RTC_RTP_MUXER, "Failed to calculate data size");
         }
-
-        for (int i = 0; i < frame_->nb_samples; i++) {
-            if (size + pcm_size * codec_ctx_->channels <= max) {
-                memcpy(buf + size,frame_->data[0] + pcm_size*codec_ctx_->channels * i, pcm_size * codec_ctx_->channels);
-                size += pcm_size * codec_ctx_->channels;
+        bool bplane = av_sample_fmt_is_planar(codec_ctx_->sample_fmt);
+        if (bplane) {
+            for (int i = 0; i < frame_->nb_samples; i++) {
+                for (int channel=0;channel< codec_ctx_->channels; channel++){
+                    if (size + pcm_size <= max) {
+                        memcpy(buf + size,frame_->data[channel] + pcm_size* i, pcm_size);
+                        size += pcm_size;// * codec_ctx_->channels;
+                    }
+                }
             }
         }
+        else{
+            int framesize = av_samples_get_buffer_size(frame_->linesize[0], frame_->channels,
+                                                       frame_->nb_samples, codec_ctx_->sample_fmt, 1);
+
+            memcpy(buf + size, frame_->data[0], framesize);
+            size += framesize;
+        }
     }

     return err;

@@ -368,7 +381,21 @@ srs_error_t SrsAudioResample::resample(SrsSample *pcm, char *buf, int &size)
     if (src_linesize_ * plane < pcm->size || pcm->size < 0) {
         return srs_error_new(ERROR_RTC_RTP_MUXER, "size not ok");
     }
-    memcpy(src_data_[0], pcm->bytes, pcm->size);
+
+    bool bsrcplane = av_sample_fmt_is_planar(src_sample_fmt_);
+    if (bsrcplane){
+        int offset = 0;
+        int src_pcm_size = av_get_bytes_per_sample(src_sample_fmt_);
+        for (int i = 0; i < src_nb_samples_; i++) {
+            for (int channel=0;channel< src_nb_channels_; channel++){
+                memcpy(src_data_[channel] + i * src_pcm_size, pcm->bytes + offset, src_pcm_size);
+                offset += src_pcm_size;
+            }
+        }
+    }
+    else{
+        memcpy(src_data_[0], pcm->bytes, pcm->size);
+    }

     dst_nb_samples_ = av_rescale_rnd(swr_get_delay(swr_ctx_, src_rate_) +
                                     src_nb_samples_, dst_rate_, src_rate_, AV_ROUND_UP);
@@ -389,16 +416,35 @@ srs_error_t SrsAudioResample::resample(SrsSample *pcm, char *buf, int &size)

     int dst_bufsize = av_samples_get_buffer_size(&dst_linesize_, dst_nb_channels_,
                                                 ret, dst_sample_fmt_, 1);
+
+
     if (dst_bufsize < 0) {
         return srs_error_new(ERROR_RTC_RTP_MUXER, "Could not get sample buffer size");
     }

-    int max = size;
-    size = 0;
-    if (max >= dst_bufsize) {
-        memcpy(buf, dst_data_[0], dst_bufsize);
-        size = dst_bufsize;
+    bool bdstplane = av_sample_fmt_is_planar(dst_sample_fmt_);
+    if (bdstplane){
+        int pcm_size = av_get_bytes_per_sample(dst_sample_fmt_);
+        int max = size;
+        size = 0;
+        for (int i = 0; i < dst_nb_samples_; i++) {
+            for (int channel=0;channel< dst_nb_channels_; channel++){
+                if (size + pcm_size <= max) {
+                    memcpy(buf + size,dst_data_[channel] + pcm_size* i, pcm_size);
+                    size += pcm_size;// * codec_ctx_->channels;
+                }
+            }
+        }
     }
+    else{
+        int max = size;
+        size = 0;
+        if (max >= dst_bufsize) {
+            memcpy(buf, dst_data_[0], dst_bufsize);
+            size = dst_bufsize;
+        }
+    }
+

     return err;
 }

TRANS_BY_GPT3

[winlinvip]

See also #2011 has not been merged yet.

TRANS_BY_GPT3

[winlinvip]

Fixed.