Index: webrtc/audio/audio_receive_stream.cc |
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc |
index 82e8ddfb2283626689b46a735c3d90e2c9a9cf8e..dd3f3c4794134cb059faf53b897f318c0236fe61 100644 |
--- a/webrtc/audio/audio_receive_stream.cc |
+++ b/webrtc/audio/audio_receive_stream.cc |
@@ -18,12 +18,14 @@ |
#include "webrtc/base/logging.h" |
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" |
#include "webrtc/system_wrappers/include/tick_util.h" |
+#include "webrtc/voice_engine/channel_proxy.h" |
#include "webrtc/voice_engine/include/voe_base.h" |
#include "webrtc/voice_engine/include/voe_codec.h" |
#include "webrtc/voice_engine/include/voe_neteq_stats.h" |
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h" |
#include "webrtc/voice_engine/include/voe_video_sync.h" |
#include "webrtc/voice_engine/include/voe_volume_control.h" |
+#include "webrtc/voice_engine/voice_engine_impl.h" |
namespace webrtc { |
std::string AudioReceiveStream::Config::Rtp::ToString() const { |
@@ -74,24 +76,26 @@ AudioReceiveStream::AudioReceiveStream( |
RTC_DCHECK(audio_state_.get()); |
RTC_DCHECK(rtp_header_parser_); |
+ VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine()); |
+ channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id); |
+ channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc); |
+ |
const int channel_id = config.voe_channel_id; |
ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine()); |
- int error = rtp->SetLocalSSRC(channel_id, config.rtp.local_ssrc); |
- RTC_DCHECK_EQ(0, error); |
for (const auto& extension : config.rtp.extensions) { |
// One-byte-extension local identifiers are in the range 1-14 inclusive. |
RTC_DCHECK_GE(extension.id, 1); |
RTC_DCHECK_LE(extension.id, 14); |
if (extension.name == RtpExtension::kAudioLevel) { |
- error = rtp->SetReceiveAudioLevelIndicationStatus(channel_id, true, |
- extension.id); |
+ int error = rtp->SetReceiveAudioLevelIndicationStatus(channel_id, true, |
+ extension.id); |
RTC_DCHECK_EQ(0, error); |
bool registered = rtp_header_parser_->RegisterRtpHeaderExtension( |
kRtpExtensionAudioLevel, extension.id); |
RTC_DCHECK(registered); |
} else if (extension.name == RtpExtension::kAbsSendTime) { |
- error = rtp->SetReceiveAbsoluteSenderTimeStatus(channel_id, true, |
- extension.id); |
+ int error = rtp->SetReceiveAbsoluteSenderTimeStatus(channel_id, true, |
+ extension.id); |
RTC_DCHECK_EQ(0, error); |
bool registered = rtp_header_parser_->RegisterRtpHeaderExtension( |
kRtpExtensionAbsoluteSendTime, extension.id); |