Bug 1595479 - Remove unnecessary StartTransmitting from ConfigureSendMediaCodec; r=ng

We don't need to start transmitting until the transceiver asks us to. Doing
this early leads to creating and tearing down an audio send stream
unnecessarily, generating extra RTCP byes. These changes make the behaviour
consistent with the VideoConduit version.

Differential Revision: https://phabricator.services.mozilla.com/D57864

--HG--
extra : moz-landing-system : lando

media/webrtc/signaling/gtest/audioconduit_unittests.cpp

@@ -89,6 +89,7 @@ TEST_F(AudioConduitTest, TestConfigureSendMediaCodec) {
   AudioCodecConfig codecConfig(114, "opus", 48000, 2, false);
   ec = mAudioConduit->ConfigureSendMediaCodec(&codecConfig);
   ASSERT_EQ(ec, kMediaConduitNoError);
+  mAudioConduit->StartTransmitting();
   {
     const webrtc::SdpAudioFormat& f =
         mCall->mAudioSendConfig.send_codec_spec->format;
@@ -128,6 +129,7 @@ TEST_F(AudioConduitTest, TestConfigureSendOpusMono) {
   AudioCodecConfig codecConfig = AudioCodecConfig(114, "opus", 48000, 1, false);
   ec = mAudioConduit->ConfigureSendMediaCodec(&codecConfig);
   ASSERT_EQ(ec, kMediaConduitNoError);
+  mAudioConduit->StartTransmitting();
   {
     const webrtc::SdpAudioFormat& f =
         mCall->mAudioSendConfig.send_codec_spec->format;
@@ -147,6 +149,7 @@ TEST_F(AudioConduitTest, TestConfigureSendOpusFEC) {
   AudioCodecConfig codecConfig = AudioCodecConfig(114, "opus", 48000, 2, true);
   ec = mAudioConduit->ConfigureSendMediaCodec(&codecConfig);
   ASSERT_EQ(ec, kMediaConduitNoError);
+  mAudioConduit->StartTransmitting();
   {
     const webrtc::SdpAudioFormat& f =
         mCall->mAudioSendConfig.send_codec_spec->format;
@@ -168,6 +171,7 @@ TEST_F(AudioConduitTest, TestConfigureSendMaxPlaybackRate) {
   codecConfig.mMaxPlaybackRate = 1234;
   ec = mAudioConduit->ConfigureSendMediaCodec(&codecConfig);
   ASSERT_EQ(ec, kMediaConduitNoError);
+  mAudioConduit->StartTransmitting();
   {
     const webrtc::SdpAudioFormat& f =
         mCall->mAudioSendConfig.send_codec_spec->format;

media/webrtc/signaling/src/media-conduit/AudioConduit.cpp

@@ -403,11 +403,6 @@ MediaConduitErrorCode WebrtcAudioConduit::ConfigureSendMediaCodec(
 
   mDtmfEnabled = codecConfig->mDtmfEnabled;
 
-  condError = StartTransmitting();
-  if (condError != kMediaConduitNoError) {
-    return condError;
-  }
-
   return kMediaConduitNoError;
 }