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Unified Diff: webrtc/media/engine/webrtcvoiceengine.cc

Issue 1670153003: Introduce struct MediaConfig, with construction-time settings. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Addressed test nit; use reference. Created 4 years, 10 months ago
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Index: webrtc/media/engine/webrtcvoiceengine.cc
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index c4e92c0024b33ad37d00b28401f64a1a376a5c1a..b6c2deaf53a2839e4c20352a6d9f8e9e7af7fd84 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -649,10 +649,12 @@ rtc::scoped_refptr<webrtc::AudioState>
return audio_state_;
}
-VoiceMediaChannel* WebRtcVoiceEngine::CreateChannel(webrtc::Call* call,
+VoiceMediaChannel* WebRtcVoiceEngine::CreateChannel(
+ webrtc::Call* call,
+ const MediaConfig& config,
const AudioOptions& options) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
- return new WebRtcVoiceMediaChannel(this, options, call);
+ return new WebRtcVoiceMediaChannel(this, config, options, call);
}
bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
@@ -1366,9 +1368,10 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
};
WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
+ const MediaConfig& config,
const AudioOptions& options,
webrtc::Call* call)
- : engine_(engine), call_(call) {
+ : VoiceMediaChannel(config), engine_(engine), call_(call) {
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel";
RTC_DCHECK(call);
engine->RegisterChannel(this);
@@ -1390,6 +1393,10 @@ WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
engine()->UnregisterChannel(this);
}
+rtc::DiffServCodePoint WebRtcVoiceMediaChannel::PreferredDscp() const {
+ return kAudioDscpValue;
+}
+
bool WebRtcVoiceMediaChannel::SetSendParameters(
const AudioSendParameters& params) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
@@ -1453,9 +1460,6 @@ bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
LOG(LS_INFO) << "Setting voice channel options: "
<< options.ToString();
- // Check if DSCP value is changed from previous.
- bool dscp_option_changed = (options_.dscp != options.dscp);
-
// We retain all of the existing options, and apply the given ones
// on top. This means there is no way to "clear" options such that
// they go back to the engine default.
@@ -1465,17 +1469,6 @@ bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
"Failed to apply engine options during channel SetOptions.";
return false;
}
-
- if (dscp_option_changed) {
- rtc::DiffServCodePoint dscp = rtc::DSCP_DEFAULT;
- if (options_.dscp.value_or(false)) {
- dscp = kAudioDscpValue;
- }
- if (MediaChannel::SetDscp(dscp) != 0) {
- LOG(LS_WARNING) << "Failed to set DSCP settings for audio channel";
- }
- }
-
LOG(LS_INFO) << "Set voice channel options. Current options: "
<< options_.ToString();
return true;
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