WebRTC: Support config the bitrate of transcoding AAC to Opus.(ossrs#…

…3448)
chundonglinlin · Apr 15, 2023 · 2ac3325 · 2ac3325
1 parent 26aabe4
commit 2ac3325
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Showing 4 changed files with 52 additions and 2 deletions.
diff --git a/trunk/conf/full.conf b/trunk/conf/full.conf
@@ -550,6 +550,11 @@ vhost rtc.vhost.srs.com {
         # Overwrite by env SRS_VHOST_RTC_KEEP_BFRAME for all vhosts.
         # default: off
         keep_bframe off;
+        # The transcode audio bitrate, for RTMP to RTC.
+        # Overwrite by env SRS_VHOST_RTC_AUDIO_BITRATE for all vhosts.
+        # [8000, 192000]
+        # default: 48000
+        opus_bitrate 48000;
         ###############################################################
         # Whether enable transmuxing RTC to RTMP.
         # Overwrite by env SRS_VHOST_RTC_RTC_TO_RTMP for all vhosts.
@@ -560,6 +565,11 @@ vhost rtc.vhost.srs.com {
         # Overwrite by env SRS_VHOST_RTC_PLI_FOR_RTMP for all vhosts.
         # Default: 6.0
         pli_for_rtmp 6.0;
+        # The transcode audio bitrate, for RTC to RTMP.
+        # Overwrite by env SRS_VHOST_RTC_AUDIO_BITRATE for all vhosts.
+        # [8000, 192000]
+        # default: 48000
+        aac_bitrate 48000;
     }
     ###############################################################
     # For transmuxing RTMP to RTC, it will impact the default values if RTC is on.

diff --git a/trunk/src/app/srs_app_config.cpp b/trunk/src/app/srs_app_config.cpp
@@ -4641,6 +4641,44 @@ bool SrsConfig::get_rtc_twcc_enabled(string vhost)
     return SRS_CONF_PERFER_TRUE(conf->arg0());
 }
 
+int SrsConfig::get_rtc_opus_bitrate(string vhost)
+{
+    SRS_OVERWRITE_BY_ENV_INT("srs.vhost.rtc.opus_bitrate"); // SRS_VHOST_RTC_OPUS_BITRATE
+
+    static int DEFAULT = 48000;
+
+    SrsConfDirective* conf = get_rtc(vhost);
+    if (!conf) {
+        return DEFAULT;
+    }
+
+    conf = conf->get("opus_bitrate");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+
+    return ::atoi(conf->arg0().c_str());
+}
+
+int SrsConfig::get_rtc_aac_bitrate(string vhost)
+{
+    SRS_OVERWRITE_BY_ENV_INT("srs.vhost.rtc.aac_bitrate"); // SRS_VHOST_RTC_AAC_BITRATE
+
+    static int DEFAULT = 48000;
+
+    SrsConfDirective* conf = get_rtc(vhost);
+    if (!conf) {
+        return DEFAULT;
+    }
+
+    conf = conf->get("aac_bitrate");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+
+    return ::atoi(conf->arg0().c_str());
+}
+
 SrsConfDirective* SrsConfig::get_vhost(string vhost, bool try_default_vhost)
 {
     srs_assert(root);

diff --git a/trunk/src/app/srs_app_config.hpp b/trunk/src/app/srs_app_config.hpp
@@ -531,6 +531,8 @@ class SrsConfig
     bool get_rtc_nack_enabled(std::string vhost);
     bool get_rtc_nack_no_copy(std::string vhost);
     bool get_rtc_twcc_enabled(std::string vhost);
+    int get_rtc_opus_bitrate(std::string vhost);
+    int get_rtc_aac_bitrate(std::string vhost);
 
 // vhost specified section
 public:

diff --git a/trunk/src/app/srs_app_rtc_source.cpp b/trunk/src/app/srs_app_rtc_source.cpp
@@ -877,7 +877,7 @@ srs_error_t SrsRtcRtpBuilder::init_codec(SrsAudioCodecId codec)
     codec_ = new SrsAudioTranscoder();
 
     // Initialize the codec according to the codec in stream.
-    int bitrate = 48000; // The output bitrate in bps.
+    int bitrate = _srs_config->get_rtc_opus_bitrate(req->vhost);// The output bitrate in bps.
     if ((err = codec_->initialize(codec, SrsAudioCodecIdOpus, kAudioChannel, kAudioSamplerate, bitrate)) != srs_success) {
         return srs_error_wrap(err, "init codec=%d", codec);
     }
@@ -1334,7 +1334,7 @@ srs_error_t SrsRtcFrameBuilder::initialize(SrsRequest* r)
     SrsAudioCodecId to = SrsAudioCodecIdAAC; // The output audio codec.
     int channels = 2; // The output audio channels.
     int sample_rate = 48000; // The output audio sample rate in HZ.
-    int bitrate = 48000; // The output audio bitrate in bps.
+    int bitrate = _srs_config->get_rtc_aac_bitrate(r->vhost); // The output audio bitrate in bps.
     if ((err = codec_->initialize(from, to, channels, sample_rate, bitrate)) != srs_success) {
         return srs_error_wrap(err, "bridge initialize");
     }