Injectable audio encoders: voice_engine/channel changes.
Adds a SetEncoder call to voe::Channel, so that we can move encoder setup outside of Voice Engine.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2703373006
Cr-Original-Commit-Position: refs/heads/master@{#17572}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 1ffbd6c93cf322de23b3772c39021c58abe4b7bc
diff --git a/test/mock_voe_channel_proxy.h b/test/mock_voe_channel_proxy.h
index b62c266..a0ccd45 100644
--- a/test/mock_voe_channel_proxy.h
+++ b/test/mock_voe_channel_proxy.h
@@ -22,6 +22,14 @@
class MockVoEChannelProxy : public voe::ChannelProxy {
public:
+ // GTest doesn't like move-only types, like std::unique_ptr
+ bool SetEncoder(int payload_type,
+ std::unique_ptr<AudioEncoder> encoder) {
+ return SetEncoderForMock(payload_type, &encoder);
+ }
+ MOCK_METHOD2(SetEncoderForMock,
+ bool(int payload_type,
+ std::unique_ptr<AudioEncoder>* encoder));
MOCK_METHOD1(SetRTCPStatus, void(bool enable));
MOCK_METHOD1(SetLocalSSRC, void(uint32_t ssrc));
MOCK_METHOD1(SetRTCP_CNAME, void(const std::string& c_name));
diff --git a/voice_engine/BUILD.gn b/voice_engine/BUILD.gn
index ba79b2d..7a4641f 100644
--- a/voice_engine/BUILD.gn
+++ b/voice_engine/BUILD.gn
@@ -146,6 +146,7 @@
"../call:call_interfaces",
"../common_audio",
"../logging:rtc_event_log_api",
+ "../modules/audio_coding:audio_encoder_interface",
"../modules/audio_coding:audio_format_conversion",
"../modules/audio_coding:rent_a_codec",
"../modules/audio_conference_mixer",
diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc
index 3a4a970..a7a45e6 100644
--- a/voice_engine/channel.cc
+++ b/voice_engine/channel.cc
@@ -1274,6 +1274,38 @@
_rtpRtcpModule->SetSendingMediaStatus(false);
}
+bool Channel::SetEncoder(int payload_type,
+ std::unique_ptr<AudioEncoder> encoder) {
+ RTC_DCHECK_GE(payload_type, 0);
+ RTC_DCHECK_LE(payload_type, 127);
+ // TODO(ossu): Make a CodecInst up for now. It seems like very little of this
+ // information is actually used, possibly only payload type and clock rate.
+ CodecInst lies;
+ lies.pltype = payload_type;
+ strncpy(lies.plname, "audio", sizeof(lies.plname));
+ lies.plname[sizeof(lies.plname) - 1] = 0;
+ // Seems unclear if it should be clock rate or sample rate. CodecInst
+ // supposedly carries the sample rate, but only clock rate seems sensible to
+ // send to the RTP/RTCP module.
+ lies.plfreq = encoder->RtpTimestampRateHz();
+ lies.pacsize = 0;
+ lies.channels = encoder->NumChannels();
+ lies.rate = 0;
+
+ if (_rtpRtcpModule->RegisterSendPayload(lies) != 0) {
+ _rtpRtcpModule->DeRegisterSendPayload(payload_type);
+ if (_rtpRtcpModule->RegisterSendPayload(lies) != 0) {
+ WEBRTC_TRACE(
+ kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
+ "SetEncoder() failed to register codec to RTP/RTCP module");
+ return false;
+ }
+ }
+
+ audio_coding_->SetEncoder(std::move(encoder));
+ return true;
+}
+
int32_t Channel::RegisterVoiceEngineObserver(VoiceEngineObserver& observer) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::RegisterVoiceEngineObserver()");
diff --git a/voice_engine/channel.h b/voice_engine/channel.h
index 13d8e49..b43c737 100644
--- a/voice_engine/channel.h
+++ b/voice_engine/channel.h
@@ -23,6 +23,7 @@
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/acm2/codec_manager.h"
#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
#include "webrtc/modules/audio_processing/rms_level.h"
@@ -172,6 +173,9 @@
void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
+ // Send using this encoder, with this payload type.
+ bool SetEncoder(int payload_type, std::unique_ptr<AudioEncoder> encoder);
+
// API methods
// VoEBase
diff --git a/voice_engine/channel_proxy.cc b/voice_engine/channel_proxy.cc
index bb56b07..ac96fe9 100644
--- a/voice_engine/channel_proxy.cc
+++ b/voice_engine/channel_proxy.cc
@@ -30,6 +30,12 @@
ChannelProxy::~ChannelProxy() {}
+bool ChannelProxy::SetEncoder(int payload_type,
+ std::unique_ptr<AudioEncoder> encoder) {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ return channel()->SetEncoder(payload_type, std::move(encoder));
+}
+
void ChannelProxy::SetRTCPStatus(bool enable) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->SetRTCPStatus(enable);
diff --git a/voice_engine/channel_proxy.h b/voice_engine/channel_proxy.h
index 86b5e65..d964bd4 100644
--- a/voice_engine/channel_proxy.h
+++ b/voice_engine/channel_proxy.h
@@ -15,6 +15,7 @@
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/race_checker.h"
#include "webrtc/base/thread_checker.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/voice_engine/channel_manager.h"
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
@@ -54,6 +55,9 @@
explicit ChannelProxy(const ChannelOwner& channel_owner);
virtual ~ChannelProxy();
+ virtual bool SetEncoder(int payload_type,
+ std::unique_ptr<AudioEncoder> encoder);
+
virtual void SetRTCPStatus(bool enable);
virtual void SetLocalSSRC(uint32_t ssrc);
virtual void SetRTCP_CNAME(const std::string& c_name);
