Remove channel ids from various interfaces.
Starts by removing channel/engine id from ViEChannel which propagates
down to the RTP/RTCP module as well as the transport class.
IncomingVideoStream::RenderFrame() is untouched for now but receives a
fake id instead of the previous channel id. Added a TODO to remove it
later but the RenderFrame call is implemented in a lot of
platform-dependent files and should probably remove the "manager" aspect
of renderers, so preferring to do it separately
BUG=webrtc:1695
R=henrika@webrtc.org, mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1335353005 .
Cr-Commit-Position: refs/heads/master@{#9978}
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index aa7d7da..5bf1899 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -333,13 +333,13 @@
VoiceMediaChannel::Error* error) override;
// implements Transport interface
- int SendPacket(int channel, const void* data, size_t len) override {
+ int SendPacket(const void* data, size_t len) override {
rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
kMaxRtpPacketLen);
return VoiceMediaChannel::SendPacket(&packet) ? static_cast<int>(len) : -1;
}
- int SendRTCPPacket(int channel, const void* data, size_t len) override {
+ int SendRTCPPacket(const void* data, size_t len) override {
rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
kMaxRtpPacketLen);
return VoiceMediaChannel::SendRtcp(&packet) ? static_cast<int>(len) : -1;
diff --git a/webrtc/common_types.h b/webrtc/common_types.h
index 059bd10..7be74e6 100644
--- a/webrtc/common_types.h
+++ b/webrtc/common_types.h
@@ -166,15 +166,14 @@
};
// External transport callback interface
-class Transport
-{
-public:
- virtual int SendPacket(int channel, const void *data, size_t len) = 0;
- virtual int SendRTCPPacket(int channel, const void *data, size_t len) = 0;
+class Transport {
+ public:
+ virtual int SendPacket(const void* data, size_t len) = 0;
+ virtual int SendRTCPPacket(const void* data, size_t len) = 0;
-protected:
- virtual ~Transport() {}
- Transport() {}
+ protected:
+ virtual ~Transport() {}
+ Transport() {}
};
// Statistics for an RTCP channel
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h b/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h
index 6283566..2fb8ac5 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h
@@ -37,14 +37,14 @@
public:
// Creates a video-enabled RTP receiver.
static RtpReceiver* CreateVideoReceiver(
- int id, Clock* clock,
+ Clock* clock,
RtpData* incoming_payload_callback,
RtpFeedback* incoming_messages_callback,
RTPPayloadRegistry* rtp_payload_registry);
// Creates an audio-enabled RTP receiver.
static RtpReceiver* CreateAudioReceiver(
- int id, Clock* clock,
+ Clock* clock,
RtpAudioFeedback* incoming_audio_feedback,
RtpData* incoming_payload_callback,
RtpFeedback* incoming_messages_callback,
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
index 15e0ffc..4dfb1dd 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
@@ -57,7 +57,6 @@
* paced_sender - Spread any bursts of packets into smaller
* bursts to minimize packet loss.
*/
- int32_t id;
bool audio;
bool receiver_only;
Clock* clock;
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
index 8f3500e..73fb96c 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
@@ -242,28 +242,24 @@
* channels - number of channels in codec (1 = mono, 2 = stereo)
*/
virtual int32_t OnInitializeDecoder(
- const int32_t id,
const int8_t payloadType,
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const int frequency,
const uint8_t channels,
const uint32_t rate) = 0;
- virtual void OnIncomingSSRCChanged( const int32_t id,
- const uint32_t ssrc) = 0;
+ virtual void OnIncomingSSRCChanged(const uint32_t ssrc) = 0;
- virtual void OnIncomingCSRCChanged( const int32_t id,
- const uint32_t CSRC,
- const bool added) = 0;
+ virtual void OnIncomingCSRCChanged(const uint32_t CSRC,
+ const bool added) = 0;
};
class RtpAudioFeedback {
public:
-
- virtual void OnPlayTelephoneEvent(const int32_t id,
- const uint8_t event,
+ virtual void OnPlayTelephoneEvent(const uint8_t event,
const uint16_t lengthMs,
const uint8_t volume) = 0;
+
protected:
virtual ~RtpAudioFeedback() {}
};
@@ -348,8 +344,7 @@
public:
virtual ~NullRtpFeedback() {}
- int32_t OnInitializeDecoder(const int32_t id,
- const int8_t payloadType,
+ int32_t OnInitializeDecoder(const int8_t payloadType,
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const int frequency,
const uint8_t channels,
@@ -357,11 +352,8 @@
return 0;
}
- void OnIncomingSSRCChanged(const int32_t id, const uint32_t ssrc) override {}
-
- void OnIncomingCSRCChanged(const int32_t id,
- const uint32_t CSRC,
- const bool added) override {}
+ void OnIncomingSSRCChanged(const uint32_t ssrc) override {}
+ void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) override {}
};
// Null object version of RtpData.
@@ -385,8 +377,7 @@
public:
virtual ~NullRtpAudioFeedback() {}
- void OnPlayTelephoneEvent(const int32_t id,
- const uint8_t event,
+ void OnPlayTelephoneEvent(const uint8_t event,
const uint16_t lengthMs,
const uint8_t volume) override {}
};
diff --git a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
index 9edf3ed..d32d09f 100644
--- a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
@@ -26,7 +26,6 @@
using namespace webrtc;
const int kVideoNackListSize = 30;
-const int kTestId = 123;
const uint32_t kTestSsrc = 3456;
const uint16_t kTestSequenceNumber = 2345;
const uint32_t kTestNumberOfPackets = 1350;
@@ -57,7 +56,7 @@
TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {}
virtual ~TestRtpFeedback() {}
- void OnIncomingSSRCChanged(const int32_t id, const uint32_t ssrc) override {
+ void OnIncomingSSRCChanged(const uint32_t ssrc) override {
rtp_rtcp_->SetRemoteSSRC(ssrc);
}
@@ -96,7 +95,7 @@
packet_loss_ = 0;
}
- int SendPacket(int channel, const void* data, size_t len) override {
+ int SendPacket(const void* data, size_t len) override {
count_++;
const unsigned char* ptr = static_cast<const unsigned char*>(data);
uint32_t ssrc = (ptr[8] << 24) + (ptr[9] << 16) + (ptr[10] << 8) + ptr[11];
@@ -155,7 +154,7 @@
return static_cast<int>(len);
}
- int SendRTCPPacket(int channel, const void* data, size_t len) override {
+ int SendRTCPPacket(const void* data, size_t len) override {
if (module_->IncomingRtcpPacket((const uint8_t*)data, len) == 0) {
return static_cast<int>(len);
}
@@ -186,7 +185,6 @@
void SetUp() override {
RtpRtcp::Configuration configuration;
- configuration.id = kTestId;
configuration.audio = false;
configuration.clock = &fake_clock;
receive_statistics_.reset(ReceiveStatistics::Create(&fake_clock));
@@ -197,7 +195,7 @@
rtp_feedback_.reset(new TestRtpFeedback(rtp_rtcp_module_));
rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver(
- kTestId, &fake_clock, &receiver_, rtp_feedback_.get(),
+ &fake_clock, &receiver_, rtp_feedback_.get(),
&rtp_payload_registry_));
rtp_rtcp_module_->SetSSRC(kTestSsrc);
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
index 394fa68..a03af24 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
@@ -31,14 +31,8 @@
rtcp_receiver_(rtcp_receiver) {
}
- int SendPacket(int /*channel*/,
- const void* /*data*/,
- size_t /*len*/) override {
- return -1;
- }
- int SendRTCPPacket(int /*channel*/,
- const void* packet,
- size_t packetLength) override {
+ int SendPacket(const void* /*data*/, size_t /*len*/) override { return -1; }
+ int SendRTCPPacket(const void* packet, size_t packetLength) override {
RTCPUtility::RTCPParserV2 rtcpParser((uint8_t*)packet,
packetLength,
true); // Allow non-compound RTCP
@@ -88,14 +82,13 @@
void RtcpFormatRembTest::SetUp() {
RtpRtcp::Configuration configuration;
- configuration.id = 0;
configuration.audio = false;
configuration.clock = system_clock_;
configuration.remote_bitrate_estimator = remote_bitrate_estimator_.get();
dummy_rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(configuration);
- rtcp_sender_ = new RTCPSender(0, false, system_clock_,
- receive_statistics_.get(), NULL);
- rtcp_receiver_ = new RTCPReceiver(0, system_clock_, false, NULL, NULL, NULL,
+ rtcp_sender_ =
+ new RTCPSender(false, system_clock_, receive_statistics_.get(), NULL);
+ rtcp_receiver_ = new RTCPReceiver(system_clock_, false, NULL, NULL, NULL,
dummy_rtp_rtcp_impl_);
test_transport_ = new TestTransport(rtcp_receiver_);
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
index f9dc96e..4392f52 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -30,7 +30,6 @@
const int kRrTimeoutIntervals = 3;
RTCPReceiver::RTCPReceiver(
- int32_t id,
Clock* clock,
bool receiver_only,
RtcpPacketTypeCounterObserver* packet_type_counter_observer,
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
index 6ae47a6..9392e51 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -29,8 +29,7 @@
class RTCPReceiver : public TMMBRHelp
{
public:
- RTCPReceiver(int32_t id,
- Clock* clock,
+ RTCPReceiver(Clock* clock,
bool receiver_only,
RtcpPacketTypeCounterObserver* packet_type_counter_observer,
RtcpBandwidthObserver* rtcp_bandwidth_observer,
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
index ce66613..6ef2a14 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
@@ -39,15 +39,12 @@
void SetRTCPReceiver(RTCPReceiver* rtcp_receiver) {
rtcp_receiver_ = rtcp_receiver;
}
- int SendPacket(int /*ch*/, const void* /*data*/, size_t /*len*/) override {
+ int SendPacket(const void* /*data*/, size_t /*len*/) override {
ADD_FAILURE(); // FAIL() gives a compile error.
return -1;
}
- // Injects an RTCP packet into the receiver.
- int SendRTCPPacket(int /* ch */,
- const void* packet,
- size_t packet_len) override {
+ int SendRTCPPacket(const void* packet, size_t packet_len) override {
ADD_FAILURE();
return 0;
}
@@ -76,14 +73,13 @@
test_transport_ = new TestTransport();
RtpRtcp::Configuration configuration;
- configuration.id = 0;
configuration.audio = false;
configuration.clock = &system_clock_;
configuration.outgoing_transport = test_transport_;
configuration.remote_bitrate_estimator = remote_bitrate_estimator_.get();
rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(configuration);
- rtcp_receiver_ = new RTCPReceiver(0, &system_clock_, false, NULL, NULL,
- NULL, rtp_rtcp_impl_);
+ rtcp_receiver_ = new RTCPReceiver(&system_clock_, false, NULL, NULL, NULL,
+ rtp_rtcp_impl_);
test_transport_->SetRTCPReceiver(rtcp_receiver_);
}
~RtcpReceiverTest() {
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
index ea7931f..8b7bbb3 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -132,13 +132,11 @@
};
RTCPSender::RTCPSender(
- int32_t id,
bool audio,
Clock* clock,
ReceiveStatistics* receive_statistics,
RtcpPacketTypeCounterObserver* packet_type_counter_observer)
- : id_(id),
- audio_(audio),
+ : audio_(audio),
clock_(clock),
method_(kRtcpOff),
critical_section_transport_(
@@ -1119,7 +1117,7 @@
int32_t RTCPSender::SendToNetwork(const uint8_t* dataBuffer, size_t length) {
CriticalSectionScoped lock(critical_section_transport_.get());
if (cbTransport_) {
- if (cbTransport_->SendRTCPPacket(id_, dataBuffer, length) > 0)
+ if (cbTransport_->SendRTCPPacket(dataBuffer, length) > 0)
return 0;
}
return -1;
@@ -1218,18 +1216,17 @@
class Sender : public rtcp::RtcpPacket::PacketReadyCallback {
public:
- Sender(Transport* transport, int32_t id)
- : transport_(transport), id_(id), send_failure_(false) {}
+ Sender(Transport* transport)
+ : transport_(transport), send_failure_(false) {}
void OnPacketReady(uint8_t* data, size_t length) override {
- if (transport_->SendRTCPPacket(id_, data, length) <= 0)
+ if (transport_->SendRTCPPacket(data, length) <= 0)
send_failure_ = true;
}
Transport* const transport_;
- int32_t id_;
bool send_failure_;
- } sender(cbTransport_, id_);
+ } sender(cbTransport_);
uint8_t buffer[IP_PACKET_SIZE];
return packet.BuildExternalBuffer(buffer, IP_PACKET_SIZE, &sender) &&
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
index 083ce78..e0195f1 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
@@ -73,8 +73,7 @@
ModuleRtpRtcpImpl* module;
};
- RTCPSender(int32_t id,
- bool audio,
+ RTCPSender(bool audio,
Clock* clock,
ReceiveStatistics* receive_statistics,
RtcpPacketTypeCounterObserver* packet_type_counter_observer);
@@ -227,7 +226,6 @@
EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
private:
- const int32_t id_;
const bool audio_;
Clock* const clock_;
RTCPMethod method_ GUARDED_BY(critical_section_rtcp_sender_);
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
index d9b975a..9a6bf11 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
@@ -199,10 +199,10 @@
public:
TestTransport() {}
- int SendPacket(int /*ch*/, const void* /*data*/, size_t /*len*/) override {
+ int SendPacket(const void* /*data*/, size_t /*len*/) override {
return -1;
}
- int SendRTCPPacket(int /*ch*/, const void* data, size_t len) override {
+ int SendRTCPPacket(const void* data, size_t len) override {
parser_.Parse(static_cast<const uint8_t*>(data), len);
return static_cast<int>(len);
}
@@ -225,14 +225,13 @@
: clock_(1335900000),
receive_statistics_(ReceiveStatistics::Create(&clock_)) {
RtpRtcp::Configuration configuration;
- configuration.id = 0;
configuration.audio = false;
configuration.clock = &clock_;
configuration.outgoing_transport = &test_transport_;
rtp_rtcp_impl_.reset(new ModuleRtpRtcpImpl(configuration));
- rtcp_sender_.reset(new RTCPSender(configuration.id, false, &clock_,
- receive_statistics_.get(), nullptr));
+ rtcp_sender_.reset(
+ new RTCPSender(false, &clock_, receive_statistics_.get(), nullptr));
rtcp_sender_->SetSSRC(kSenderSsrc);
rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
EXPECT_EQ(0, rtcp_sender_->RegisterSendTransport(&test_transport_));
@@ -669,7 +668,7 @@
TEST_F(RtcpSenderTest, TestRegisterRtcpPacketTypeObserver) {
RtcpPacketTypeCounterObserverImpl observer;
rtcp_sender_.reset(
- new RTCPSender(0, false, &clock_, receive_statistics_.get(), &observer));
+ new RTCPSender(false, &clock_, receive_statistics_.get(), &observer));
rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
EXPECT_EQ(0, rtcp_sender_->RegisterSendTransport(&test_transport_));
rtcp_sender_->SetRTCPStatus(kRtcpNonCompound);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
index c9a1adf..d7bf405 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
@@ -20,17 +20,15 @@
namespace webrtc {
RTPReceiverStrategy* RTPReceiverStrategy::CreateAudioStrategy(
- int32_t id, RtpData* data_callback,
+ RtpData* data_callback,
RtpAudioFeedback* incoming_messages_callback) {
- return new RTPReceiverAudio(id, data_callback, incoming_messages_callback);
+ return new RTPReceiverAudio(data_callback, incoming_messages_callback);
}
-RTPReceiverAudio::RTPReceiverAudio(const int32_t id,
- RtpData* data_callback,
+RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback,
RtpAudioFeedback* incoming_messages_callback)
: RTPReceiverStrategy(data_callback),
TelephoneEventHandler(),
- id_(id),
last_received_frequency_(8000),
telephone_event_forward_to_decoder_(false),
telephone_event_payload_type_(-1),
@@ -263,16 +261,13 @@
int32_t RTPReceiverAudio::InvokeOnInitializeDecoder(
RtpFeedback* callback,
- int32_t id,
int8_t payload_type,
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const PayloadUnion& specific_payload) const {
- if (-1 == callback->OnInitializeDecoder(id,
- payload_type,
- payload_name,
- specific_payload.Audio.frequency,
- specific_payload.Audio.channels,
- specific_payload.Audio.rate)) {
+ if (-1 ==
+ callback->OnInitializeDecoder(
+ payload_type, payload_name, specific_payload.Audio.frequency,
+ specific_payload.Audio.channels, specific_payload.Audio.rate)) {
LOG(LS_ERROR) << "Failed to create decoder for payload type: "
<< payload_name << "/" << static_cast<int>(payload_type);
return -1;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
index a7efcbb..176852e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
@@ -28,8 +28,7 @@
class RTPReceiverAudio : public RTPReceiverStrategy,
public TelephoneEventHandler {
public:
- RTPReceiverAudio(const int32_t id,
- RtpData* data_callback,
+ RTPReceiverAudio(RtpData* data_callback,
RtpAudioFeedback* incoming_messages_callback);
virtual ~RTPReceiverAudio() {}
@@ -74,7 +73,6 @@
int32_t InvokeOnInitializeDecoder(
RtpFeedback* callback,
- int32_t id,
int8_t payload_type,
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const PayloadUnion& specific_payload) const override;
@@ -106,8 +104,6 @@
const AudioPayload& audio_specific,
bool is_red);
- int32_t id_;
-
uint32_t last_received_frequency_;
bool telephone_event_forward_to_decoder_;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
index 6be0c5a..40612a6 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
@@ -27,7 +27,7 @@
using RtpUtility::StringCompare;
RtpReceiver* RtpReceiver::CreateVideoReceiver(
- int id, Clock* clock,
+ Clock* clock,
RtpData* incoming_payload_callback,
RtpFeedback* incoming_messages_callback,
RTPPayloadRegistry* rtp_payload_registry) {
@@ -36,13 +36,13 @@
if (!incoming_messages_callback)
incoming_messages_callback = NullObjectRtpFeedback();
return new RtpReceiverImpl(
- id, clock, NullObjectRtpAudioFeedback(), incoming_messages_callback,
+ clock, NullObjectRtpAudioFeedback(), incoming_messages_callback,
rtp_payload_registry,
RTPReceiverStrategy::CreateVideoStrategy(incoming_payload_callback));
}
RtpReceiver* RtpReceiver::CreateAudioReceiver(
- int id, Clock* clock,
+ Clock* clock,
RtpAudioFeedback* incoming_audio_feedback,
RtpData* incoming_payload_callback,
RtpFeedback* incoming_messages_callback,
@@ -54,25 +54,24 @@
if (!incoming_messages_callback)
incoming_messages_callback = NullObjectRtpFeedback();
return new RtpReceiverImpl(
- id, clock, incoming_audio_feedback, incoming_messages_callback,
+ clock, incoming_audio_feedback, incoming_messages_callback,
rtp_payload_registry,
- RTPReceiverStrategy::CreateAudioStrategy(id, incoming_payload_callback,
+ RTPReceiverStrategy::CreateAudioStrategy(incoming_payload_callback,
incoming_audio_feedback));
}
-RtpReceiverImpl::RtpReceiverImpl(int32_t id,
- Clock* clock,
- RtpAudioFeedback* incoming_audio_messages_callback,
- RtpFeedback* incoming_messages_callback,
- RTPPayloadRegistry* rtp_payload_registry,
- RTPReceiverStrategy* rtp_media_receiver)
+RtpReceiverImpl::RtpReceiverImpl(
+ Clock* clock,
+ RtpAudioFeedback* incoming_audio_messages_callback,
+ RtpFeedback* incoming_messages_callback,
+ RTPPayloadRegistry* rtp_payload_registry,
+ RTPReceiverStrategy* rtp_media_receiver)
: clock_(clock),
rtp_payload_registry_(rtp_payload_registry),
rtp_media_receiver_(rtp_media_receiver),
- id_(id),
cb_rtp_feedback_(incoming_messages_callback),
critical_section_rtp_receiver_(
- CriticalSectionWrapper::CreateCriticalSection()),
+ CriticalSectionWrapper::CreateCriticalSection()),
last_receive_time_(0),
last_received_payload_length_(0),
ssrc_(0),
@@ -90,8 +89,7 @@
RtpReceiverImpl::~RtpReceiverImpl() {
for (int i = 0; i < num_csrcs_; ++i) {
- cb_rtp_feedback_->OnIncomingCSRCChanged(id_, current_remote_csrc_[i],
- false);
+ cb_rtp_feedback_->OnIncomingCSRCChanged(current_remote_csrc_[i], false);
}
}
@@ -299,13 +297,14 @@
if (new_ssrc) {
// We need to get this to our RTCP sender and receiver.
// We need to do this outside critical section.
- cb_rtp_feedback_->OnIncomingSSRCChanged(id_, rtp_header.ssrc);
+ cb_rtp_feedback_->OnIncomingSSRCChanged(rtp_header.ssrc);
}
if (re_initialize_decoder) {
- if (-1 == cb_rtp_feedback_->OnInitializeDecoder(
- id_, rtp_header.payloadType, payload_name,
- rtp_header.payload_type_frequency, channels, rate)) {
+ if (-1 ==
+ cb_rtp_feedback_->OnInitializeDecoder(
+ rtp_header.payloadType, payload_name,
+ rtp_header.payload_type_frequency, channels, rate)) {
// New stream, same codec.
LOG(LS_ERROR) << "Failed to create decoder for payload type: "
<< static_cast<int>(rtp_header.payloadType);
@@ -397,9 +396,9 @@
} // End critsect.
if (re_initialize_decoder) {
- if (-1 == rtp_media_receiver_->InvokeOnInitializeDecoder(
- cb_rtp_feedback_, id_, payload_type, payload_name,
- *specific_payload)) {
+ if (-1 ==
+ rtp_media_receiver_->InvokeOnInitializeDecoder(
+ cb_rtp_feedback_, payload_type, payload_name, *specific_payload)) {
return -1; // Wrong payload type.
}
}
@@ -456,7 +455,7 @@
if (!found_match && csrc) {
// Didn't find it, report it as new.
have_called_callback = true;
- cb_rtp_feedback_->OnIncomingCSRCChanged(id_, csrc, true);
+ cb_rtp_feedback_->OnIncomingCSRCChanged(csrc, true);
}
}
// Search for old CSRC in new array.
@@ -473,7 +472,7 @@
if (!found_match && csrc) {
// Did not find it, report as removed.
have_called_callback = true;
- cb_rtp_feedback_->OnIncomingCSRCChanged(id_, csrc, false);
+ cb_rtp_feedback_->OnIncomingCSRCChanged(csrc, false);
}
}
if (!have_called_callback) {
@@ -481,9 +480,9 @@
// Using CSRC 0 to signal this event, not interop safe, other
// implementations might have CSRC 0 as a valid value.
if (num_csrcs_diff > 0) {
- cb_rtp_feedback_->OnIncomingCSRCChanged(id_, 0, true);
+ cb_rtp_feedback_->OnIncomingCSRCChanged(0, true);
} else if (num_csrcs_diff < 0) {
- cb_rtp_feedback_->OnIncomingCSRCChanged(id_, 0, false);
+ cb_rtp_feedback_->OnIncomingCSRCChanged(0, false);
}
}
}
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
index c904e1f..d6fbf2e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
@@ -25,8 +25,7 @@
// Callbacks passed in here may not be NULL (use Null Object callbacks if you
// want callbacks to do nothing). This class takes ownership of the media
// receiver but nothing else.
- RtpReceiverImpl(int32_t id,
- Clock* clock,
+ RtpReceiverImpl(Clock* clock,
RtpAudioFeedback* incoming_audio_messages_callback,
RtpFeedback* incoming_messages_callback,
RTPPayloadRegistry* rtp_payload_registry,
@@ -79,8 +78,6 @@
RTPPayloadRegistry* rtp_payload_registry_;
rtc::scoped_ptr<RTPReceiverStrategy> rtp_media_receiver_;
- int32_t id_;
-
RtpFeedback* cb_rtp_feedback_;
rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtp_receiver_;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
index 9c09f8e..a9e85ec 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
@@ -28,7 +28,7 @@
public:
static RTPReceiverStrategy* CreateVideoStrategy(RtpData* data_callback);
static RTPReceiverStrategy* CreateAudioStrategy(
- int32_t id, RtpData* data_callback,
+ RtpData* data_callback,
RtpAudioFeedback* incoming_messages_callback);
virtual ~RTPReceiverStrategy() {}
@@ -70,7 +70,6 @@
// Invokes the OnInitializeDecoder callback in a media-specific way.
virtual int32_t InvokeOnInitializeDecoder(
RtpFeedback* callback,
- int32_t id,
int8_t payload_type,
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const PayloadUnion& specific_payload) const = 0;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
index 7537d8e..a8db0d2 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
@@ -111,14 +111,13 @@
int32_t RTPReceiverVideo::InvokeOnInitializeDecoder(
RtpFeedback* callback,
- int32_t id,
int8_t payload_type,
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const PayloadUnion& specific_payload) const {
// For video we just go with default values.
if (-1 ==
- callback->OnInitializeDecoder(
- id, payload_type, payload_name, kVideoPayloadTypeFrequency, 1, 0)) {
+ callback->OnInitializeDecoder(payload_type, payload_name,
+ kVideoPayloadTypeFrequency, 1, 0)) {
LOG(LS_ERROR) << "Failed to created decoder for payload type: "
<< static_cast<int>(payload_type);
return -1;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
index 8528a7d..23128df 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
@@ -49,7 +49,6 @@
int32_t InvokeOnInitializeDecoder(
RtpFeedback* callback,
- int32_t id,
int8_t payload_type,
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const PayloadUnion& specific_payload) const override;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 451360a..b245e81 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -27,8 +27,7 @@
namespace webrtc {
RtpRtcp::Configuration::Configuration()
- : id(-1),
- audio(false),
+ : audio(false),
receiver_only(false),
clock(nullptr),
receive_statistics(NullObjectReceiveStatistics()),
@@ -60,8 +59,7 @@
}
ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
- : rtp_sender_(configuration.id,
- configuration.audio,
+ : rtp_sender_(configuration.audio,
configuration.clock,
configuration.outgoing_transport,
configuration.audio_messages,
@@ -71,20 +69,17 @@
configuration.send_bitrate_observer,
configuration.send_frame_count_observer,
configuration.send_side_delay_observer),
- rtcp_sender_(configuration.id,
- configuration.audio,
+ rtcp_sender_(configuration.audio,
configuration.clock,
configuration.receive_statistics,
configuration.rtcp_packet_type_counter_observer),
- rtcp_receiver_(configuration.id,
- configuration.clock,
+ rtcp_receiver_(configuration.clock,
configuration.receiver_only,
configuration.rtcp_packet_type_counter_observer,
configuration.bandwidth_callback,
configuration.intra_frame_callback,
this),
clock_(configuration.clock),
- id_(configuration.id),
audio_(configuration.audio),
collision_detected_(false),
last_process_time_(configuration.clock->TimeInMilliseconds()),
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index fe69437..11403a0 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -353,7 +353,6 @@
bool TimeToSendFullNackList(int64_t now) const;
- int32_t id_;
const bool audio_;
bool collision_detected_;
int64_t last_process_time_;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
index 12630f7..881985d 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
@@ -62,7 +62,7 @@
clock_ = clock;
delay_ms_ = delay_ms;
}
- int SendPacket(int /*ch*/, const void* data, size_t len) override {
+ int SendPacket(const void* data, size_t len) override {
RTPHeader header;
rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
EXPECT_TRUE(parser->Parse(static_cast<const uint8_t*>(data), len, &header));
@@ -70,7 +70,7 @@
last_rtp_header_ = header;
return static_cast<int>(len);
}
- int SendRTCPPacket(int /*ch*/, const void* data, size_t len) override {
+ int SendRTCPPacket(const void* data, size_t len) override {
test::RtcpPacketParser parser;
parser.Parse(static_cast<const uint8_t*>(data), len);
last_nack_list_ = parser.nack_item()->last_nack_list();
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 8e1f77a..f683f14 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -97,8 +97,7 @@
uint32_t ssrc_;
};
-RTPSender::RTPSender(int32_t id,
- bool audio,
+RTPSender::RTPSender(bool audio,
Clock* clock,
Transport* transport,
RtpAudioFeedback* audio_feedback,
@@ -115,10 +114,8 @@
TickTime::MillisecondTimestamp()),
bitrates_(new BitrateAggregator(bitrate_callback)),
total_bitrate_sent_(clock, bitrates_->total_bitrate_observer()),
- id_(id),
audio_configured_(audio),
- audio_(audio ? new RTPSenderAudio(id, clock, this, audio_feedback)
- : nullptr),
+ audio_(audio ? new RTPSenderAudio(clock, this, audio_feedback) : nullptr),
video_(audio ? nullptr : new RTPSenderVideo(clock, this)),
paced_sender_(paced_sender),
packet_router_(packet_router),
@@ -740,7 +737,7 @@
bool RTPSender::SendPacketToNetwork(const uint8_t *packet, size_t size) {
int bytes_sent = -1;
if (transport_) {
- bytes_sent = transport_->SendPacket(id_, packet, size);
+ bytes_sent = transport_->SendPacket(packet, size);
}
TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
"RTPSender::SendPacketToNetwork", "size", size, "sent",
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index 6d11e80..f10cb75 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -86,8 +86,7 @@
class RTPSender : public RTPSenderInterface {
public:
- RTPSender(int32_t id,
- bool audio,
+ RTPSender(bool audio,
Clock* clock,
Transport* transport,
RtpAudioFeedback* audio_feedback,
@@ -388,8 +387,6 @@
rtc::scoped_ptr<BitrateAggregator> bitrates_;
Bitrate total_bitrate_sent_;
- int32_t id_;
-
const bool audio_configured_;
rtc::scoped_ptr<RTPSenderAudio> audio_;
rtc::scoped_ptr<RTPSenderVideo> video_;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
index de728f0..2f3faf5 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -20,34 +20,31 @@
static const int kDtmfFrequencyHz = 8000;
-RTPSenderAudio::RTPSenderAudio(const int32_t id,
- Clock* clock,
+RTPSenderAudio::RTPSenderAudio(Clock* clock,
RTPSender* rtpSender,
- RtpAudioFeedback* audio_feedback) :
- _id(id),
- _clock(clock),
- _rtpSender(rtpSender),
- _audioFeedback(audio_feedback),
- _sendAudioCritsect(CriticalSectionWrapper::CreateCriticalSection()),
- _packetSizeSamples(160),
- _dtmfEventIsOn(false),
- _dtmfEventFirstPacketSent(false),
- _dtmfPayloadType(-1),
- _dtmfTimestamp(0),
- _dtmfKey(0),
- _dtmfLengthSamples(0),
- _dtmfLevel(0),
- _dtmfTimeLastSent(0),
- _dtmfTimestampLastSent(0),
- _REDPayloadType(-1),
- _inbandVADactive(false),
- _cngNBPayloadType(-1),
- _cngWBPayloadType(-1),
- _cngSWBPayloadType(-1),
- _cngFBPayloadType(-1),
- _lastPayloadType(-1),
- _audioLevel_dBov(0) {
-}
+ RtpAudioFeedback* audio_feedback)
+ : _clock(clock),
+ _rtpSender(rtpSender),
+ _audioFeedback(audio_feedback),
+ _sendAudioCritsect(CriticalSectionWrapper::CreateCriticalSection()),
+ _packetSizeSamples(160),
+ _dtmfEventIsOn(false),
+ _dtmfEventFirstPacketSent(false),
+ _dtmfPayloadType(-1),
+ _dtmfTimestamp(0),
+ _dtmfKey(0),
+ _dtmfLengthSamples(0),
+ _dtmfLevel(0),
+ _dtmfTimeLastSent(0),
+ _dtmfTimestampLastSent(0),
+ _REDPayloadType(-1),
+ _inbandVADactive(false),
+ _cngNBPayloadType(-1),
+ _cngWBPayloadType(-1),
+ _cngSWBPayloadType(-1),
+ _cngFBPayloadType(-1),
+ _lastPayloadType(-1),
+ _audioLevel_dBov(0) {}
RTPSenderAudio::~RTPSenderAudio() {
}
@@ -204,7 +201,7 @@
}
if (dtmfToneStarted) {
if (_audioFeedback)
- _audioFeedback->OnPlayTelephoneEvent(_id, key, dtmfLengthMS, _dtmfLevel);
+ _audioFeedback->OnPlayTelephoneEvent(key, dtmfLengthMS, _dtmfLevel);
}
// A source MAY send events and coded audio packets for the same time
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
index 762668a..dd16fe5 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -22,10 +22,9 @@
class RTPSenderAudio: public DTMFqueue
{
public:
- RTPSenderAudio(const int32_t id,
- Clock* clock,
- RTPSender* rtpSender,
- RtpAudioFeedback* audio_feedback);
+ RTPSenderAudio(Clock* clock,
+ RTPSender* rtpSender,
+ RtpAudioFeedback* audio_feedback);
virtual ~RTPSenderAudio();
int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
@@ -73,7 +72,6 @@
const int8_t payloadType);
private:
- const int32_t _id;
Clock* const _clock;
RTPSender* const _rtpSender;
RtpAudioFeedback* const _audioFeedback;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 409be1a6..9086ea1 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -78,7 +78,7 @@
~LoopbackTransportTest() {
STLDeleteContainerPointers(sent_packets_.begin(), sent_packets_.end());
}
- int SendPacket(int channel, const void *data, size_t len) override {
+ int SendPacket(const void *data, size_t len) override {
packets_sent_++;
rtc::Buffer* buffer =
new rtc::Buffer(reinterpret_cast<const uint8_t*>(data), len);
@@ -88,7 +88,7 @@
sent_packets_.push_back(buffer);
return static_cast<int>(len);
}
- int SendRTCPPacket(int channel, const void* data, size_t len) override {
+ int SendRTCPPacket(const void* data, size_t len) override {
return -1;
}
int packets_sent_;
@@ -114,9 +114,9 @@
}
void SetUp() override {
- rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_,
- nullptr, &mock_paced_sender_, nullptr,
- nullptr, nullptr, nullptr, nullptr));
+ rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
+ &mock_paced_sender_, nullptr, nullptr,
+ nullptr, nullptr, nullptr));
rtp_sender_->SetSequenceNumber(kSeqNum);
}
@@ -826,7 +826,7 @@
TEST_F(RtpSenderTest, SendRedundantPayloads) {
MockTransport transport;
- rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport, nullptr,
+ rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport, nullptr,
&mock_paced_sender_, nullptr, nullptr,
nullptr, nullptr, nullptr));
rtp_sender_->SetSequenceNumber(kSeqNum);
@@ -862,28 +862,26 @@
// Send 10 packets of increasing size.
for (size_t i = 0; i < kNumPayloadSizes; ++i) {
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
- EXPECT_CALL(transport, SendPacket(_, _, _))
- .WillOnce(testing::ReturnArg<2>());
+ EXPECT_CALL(transport, SendPacket(_, _)).WillOnce(testing::ReturnArg<1>());
SendPacket(capture_time_ms, kPayloadSizes[i]);
rtp_sender_->TimeToSendPacket(seq_num++, capture_time_ms, false);
fake_clock_.AdvanceTimeMilliseconds(33);
}
// The amount of padding to send it too small to send a payload packet.
- EXPECT_CALL(transport,
- SendPacket(_, _, kMaxPaddingSize + rtp_header_len))
- .WillOnce(testing::ReturnArg<2>());
+ EXPECT_CALL(transport, SendPacket(_, kMaxPaddingSize + rtp_header_len))
+ .WillOnce(testing::ReturnArg<1>());
EXPECT_EQ(kMaxPaddingSize, rtp_sender_->TimeToSendPadding(49));
- EXPECT_CALL(transport, SendPacket(_, _, kPayloadSizes[0] +
- rtp_header_len + kRtxHeaderSize))
- .WillOnce(testing::ReturnArg<2>());
+ EXPECT_CALL(transport,
+ SendPacket(_, kPayloadSizes[0] + rtp_header_len + kRtxHeaderSize))
+ .WillOnce(testing::ReturnArg<1>());
EXPECT_EQ(kPayloadSizes[0], rtp_sender_->TimeToSendPadding(500));
- EXPECT_CALL(transport, SendPacket(_, _, kPayloadSizes[kNumPayloadSizes - 1] +
- rtp_header_len + kRtxHeaderSize))
- .WillOnce(testing::ReturnArg<2>());
- EXPECT_CALL(transport, SendPacket(_, _, kMaxPaddingSize + rtp_header_len))
- .WillOnce(testing::ReturnArg<2>());
+ EXPECT_CALL(transport, SendPacket(_, kPayloadSizes[kNumPayloadSizes - 1] +
+ rtp_header_len + kRtxHeaderSize))
+ .WillOnce(testing::ReturnArg<1>());
+ EXPECT_CALL(transport, SendPacket(_, kMaxPaddingSize + rtp_header_len))
+ .WillOnce(testing::ReturnArg<1>());
EXPECT_EQ(kPayloadSizes[kNumPayloadSizes - 1] + kMaxPaddingSize,
rtp_sender_->TimeToSendPadding(999));
}
@@ -960,7 +958,7 @@
FrameCounts frame_counts_;
} callback;
- rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, nullptr,
+ rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
&mock_paced_sender_, nullptr, nullptr,
nullptr, &callback, nullptr));
@@ -1013,7 +1011,7 @@
BitrateStatistics total_stats_;
BitrateStatistics retransmit_stats_;
} callback;
- rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, nullptr,
+ rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
&mock_paced_sender_, nullptr, nullptr,
&callback, nullptr, nullptr));
@@ -1072,7 +1070,7 @@
void SetUp() override {
payload_ = kAudioPayload;
- rtp_sender_.reset(new RTPSender(0, true, &fake_clock_, &transport_, nullptr,
+ rtp_sender_.reset(new RTPSender(true, &fake_clock_, &transport_, nullptr,
&mock_paced_sender_, nullptr, nullptr,
nullptr, nullptr, nullptr));
rtp_sender_->SetSequenceNumber(kSeqNum);
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
index bba1065..5b43f00 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
@@ -30,7 +30,7 @@
packet_loss_ = n;
}
-int LoopBackTransport::SendPacket(int channel, const void* data, size_t len) {
+int LoopBackTransport::SendPacket(const void* data, size_t len) {
count_++;
if (packet_loss_ > 0) {
if ((count_ % packet_loss_) == 0) {
@@ -56,9 +56,7 @@
return len;
}
-int LoopBackTransport::SendRTCPPacket(int channel,
- const void* data,
- size_t len) {
+int LoopBackTransport::SendRTCPPacket(const void* data, size_t len) {
if (rtp_rtcp_module_->IncomingRtcpPacket((const uint8_t*)data, len) < 0) {
return -1;
}
@@ -82,7 +80,6 @@
RtpRtcpAPITest() : fake_clock_(123456) {
test_csrcs_.push_back(1234);
test_csrcs_.push_back(2345);
- test_id = 123;
test_ssrc_ = 3456;
test_timestamp_ = 4567;
test_sequence_number_ = 2345;
@@ -91,17 +88,15 @@
void SetUp() override {
RtpRtcp::Configuration configuration;
- configuration.id = test_id;
configuration.audio = true;
configuration.clock = &fake_clock_;
module_.reset(RtpRtcp::CreateRtpRtcp(configuration));
rtp_payload_registry_.reset(new RTPPayloadRegistry(
RTPPayloadStrategy::CreateStrategy(true)));
rtp_receiver_.reset(RtpReceiver::CreateAudioReceiver(
- test_id, &fake_clock_, NULL, NULL, NULL, rtp_payload_registry_.get()));
+ &fake_clock_, NULL, NULL, NULL, rtp_payload_registry_.get()));
}
- int test_id;
rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
rtc::scoped_ptr<RtpReceiver> rtp_receiver_;
rtc::scoped_ptr<RtpRtcp> module_;
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
index 069cdc7..0ec1cfb 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
@@ -35,8 +35,8 @@
RtpReceiver* receiver,
ReceiveStatistics* receive_statistics);
void DropEveryNthPacket(int n);
- int SendPacket(int channel, const void* data, size_t len) override;
- int SendRTCPPacket(int channel, const void* data, size_t len) override;
+ int SendPacket(const void* data, size_t len) override;
+ int SendRTCPPacket(const void* data, size_t len) override;
private:
int count_;
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
index 61923aa..745386d 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
@@ -61,8 +61,7 @@
class RTPCallback : public NullRtpFeedback {
public:
- int32_t OnInitializeDecoder(const int32_t id,
- const int8_t payloadType,
+ int32_t OnInitializeDecoder(const int8_t payloadType,
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const int frequency,
const uint8_t channels,
@@ -80,7 +79,6 @@
RtpRtcpAudioTest() : fake_clock(123456) {
test_CSRC[0] = 1234;
test_CSRC[2] = 2345;
- test_id = 123;
test_ssrc = 3456;
test_timestamp = 4567;
test_sequence_number = 2345;
@@ -104,7 +102,6 @@
RTPPayloadStrategy::CreateStrategy(true)));
RtpRtcp::Configuration configuration;
- configuration.id = test_id;
configuration.audio = true;
configuration.clock = &fake_clock;
configuration.receive_statistics = receive_statistics1_.get();
@@ -113,18 +110,17 @@
module1 = RtpRtcp::CreateRtpRtcp(configuration);
rtp_receiver1_.reset(RtpReceiver::CreateAudioReceiver(
- test_id, &fake_clock, audioFeedback, data_receiver1, NULL,
+ &fake_clock, audioFeedback, data_receiver1, NULL,
rtp_payload_registry1_.get()));
- configuration.id = test_id + 1;
configuration.receive_statistics = receive_statistics2_.get();
configuration.outgoing_transport = transport2;
configuration.audio_messages = audioFeedback;
module2 = RtpRtcp::CreateRtpRtcp(configuration);
rtp_receiver2_.reset(RtpReceiver::CreateAudioReceiver(
- test_id + 1, &fake_clock, audioFeedback, data_receiver2, NULL,
- rtp_payload_registry2_.get()));
+ &fake_clock, audioFeedback, data_receiver2, NULL,
+ rtp_payload_registry2_.get()));
transport1->SetSendModule(module2, rtp_payload_registry2_.get(),
rtp_receiver2_.get(), receive_statistics2_.get());
@@ -143,7 +139,6 @@
delete rtp_callback;
}
- int test_id;
RtpRtcp* module1;
RtpRtcp* module2;
rtc::scoped_ptr<ReceiveStatistics> receive_statistics1_;
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
index 4fc89a9..e1d5e07 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
@@ -55,8 +55,7 @@
TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {}
virtual ~TestRtpFeedback() {}
- virtual void OnIncomingSSRCChanged(const int32_t id,
- const uint32_t ssrc) {
+ void OnIncomingSSRCChanged(const uint32_t ssrc) override {
rtp_rtcp_->SetRemoteSSRC(ssrc);
}
@@ -69,7 +68,6 @@
RtpRtcpRtcpTest() : fake_clock(123456) {
test_csrcs.push_back(1234);
test_csrcs.push_back(2345);
- test_id = 123;
test_ssrc = 3456;
test_timestamp = 4567;
test_sequence_number = 2345;
@@ -87,7 +85,6 @@
receive_statistics2_.reset(ReceiveStatistics::Create(&fake_clock));
RtpRtcp::Configuration configuration;
- configuration.id = test_id;
configuration.audio = true;
configuration.clock = &fake_clock;
configuration.receive_statistics = receive_statistics1_.get();
@@ -104,11 +101,10 @@
rtp_feedback1_.reset(new TestRtpFeedback(module1));
rtp_receiver1_.reset(RtpReceiver::CreateAudioReceiver(
- test_id, &fake_clock, NULL, receiver, rtp_feedback1_.get(),
+ &fake_clock, NULL, receiver, rtp_feedback1_.get(),
rtp_payload_registry1_.get()));
configuration.receive_statistics = receive_statistics2_.get();
- configuration.id = test_id + 1;
configuration.outgoing_transport = transport2;
configuration.intra_frame_callback = myRTCPFeedback2;
@@ -117,7 +113,7 @@
rtp_feedback2_.reset(new TestRtpFeedback(module2));
rtp_receiver2_.reset(RtpReceiver::CreateAudioReceiver(
- test_id + 1, &fake_clock, NULL, receiver, rtp_feedback2_.get(),
+ &fake_clock, NULL, receiver, rtp_feedback2_.get(),
rtp_payload_registry2_.get()));
transport1->SetSendModule(module2, rtp_payload_registry2_.get(),
@@ -179,7 +175,6 @@
delete receiver;
}
- int test_id;
rtc::scoped_ptr<TestRtpFeedback> rtp_feedback1_;
rtc::scoped_ptr<TestRtpFeedback> rtp_feedback2_;
rtc::scoped_ptr<ReceiveStatistics> receive_statistics1_;
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
index e28d5ce..ead8368 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
@@ -33,13 +33,11 @@
class RtpRtcpVideoTest : public ::testing::Test {
protected:
RtpRtcpVideoTest()
- : test_id_(123),
- rtp_payload_registry_(RTPPayloadStrategy::CreateStrategy(false)),
+ : rtp_payload_registry_(RTPPayloadStrategy::CreateStrategy(false)),
test_ssrc_(3456),
test_timestamp_(4567),
test_sequence_number_(2345),
- fake_clock(123456) {
- }
+ fake_clock(123456) {}
~RtpRtcpVideoTest() {}
virtual void SetUp() {
@@ -47,14 +45,13 @@
receiver_ = new TestRtpReceiver();
receive_statistics_.reset(ReceiveStatistics::Create(&fake_clock));
RtpRtcp::Configuration configuration;
- configuration.id = test_id_;
configuration.audio = false;
configuration.clock = &fake_clock;
configuration.outgoing_transport = transport_;
video_module_ = RtpRtcp::CreateRtpRtcp(configuration);
rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver(
- test_id_, &fake_clock, receiver_, NULL, &rtp_payload_registry_));
+ &fake_clock, receiver_, NULL, &rtp_payload_registry_));
video_module_->SetRTCPStatus(kRtcpCompound);
video_module_->SetSSRC(test_ssrc_);
diff --git a/webrtc/modules/video_coding/main/test/rtp_player.cc b/webrtc/modules/video_coding/main/test/rtp_player.cc
index e53a4ab..74a5b95 100644
--- a/webrtc/modules/video_coding/main/test/rtp_player.cc
+++ b/webrtc/modules/video_coding/main/test/rtp_player.cc
@@ -217,11 +217,10 @@
RtpRtcp::Configuration configuration;
configuration.clock = clock;
- configuration.id = 1;
configuration.audio = false;
handler->rtp_module_.reset(RtpReceiver::CreateVideoReceiver(
- configuration.id, configuration.clock, handler->payload_sink_.get(),
- NULL, handler->rtp_payload_registry_.get()));
+ configuration.clock, handler->payload_sink_.get(), NULL,
+ handler->rtp_payload_registry_.get()));
if (handler->rtp_module_.get() == NULL) {
return -1;
}
diff --git a/webrtc/test/channel_transport/udp_transport_impl.cc b/webrtc/test/channel_transport/udp_transport_impl.cc
index ab7a1ff..e16c742 100644
--- a/webrtc/test/channel_transport/udp_transport_impl.cc
+++ b/webrtc/test/channel_transport/udp_transport_impl.cc
@@ -1931,10 +1931,7 @@
return -1;
}
-int UdpTransportImpl::SendPacket(int /*channel*/,
- const void* data,
- size_t length)
-{
+int UdpTransportImpl::SendPacket(const void* data, size_t length) {
WEBRTC_TRACE(kTraceStream, kTraceTransport, _id, "%s", __FUNCTION__);
CriticalSectionScoped cs(_crit);
@@ -2000,10 +1997,7 @@
return -1;
}
-int UdpTransportImpl::SendRTCPPacket(int /*channel*/, const void* data,
- size_t length)
-{
-
+int UdpTransportImpl::SendRTCPPacket(const void* data, size_t length) {
CriticalSectionScoped cs(_crit);
if(_destIP[0] == 0)
{
diff --git a/webrtc/test/channel_transport/udp_transport_impl.h b/webrtc/test/channel_transport/udp_transport_impl.h
index 5dbf5d8..cbc53bc 100644
--- a/webrtc/test/channel_transport/udp_transport_impl.h
+++ b/webrtc/test/channel_transport/udp_transport_impl.h
@@ -116,8 +116,8 @@
size_t length,
uint16_t rtcpPort) override;
// Transport functions
- int SendPacket(int channel, const void* data, size_t length) override;
- int SendRTCPPacket(int channel, const void* data, size_t length) override;
+ int SendPacket(const void* data, size_t length) override;
+ int SendRTCPPacket(const void* data, size_t length) override;
// UdpTransport functions continue.
int32_t SetSendIP(const char* ipaddr) override;
diff --git a/webrtc/test/mock_transport.h b/webrtc/test/mock_transport.h
index 5b1cb8d..823baf4 100644
--- a/webrtc/test/mock_transport.h
+++ b/webrtc/test/mock_transport.h
@@ -18,10 +18,8 @@
class MockTransport : public webrtc::Transport {
public:
- MOCK_METHOD3(SendPacket,
- int(int channel, const void* data, size_t len));
- MOCK_METHOD3(SendRTCPPacket,
- int(int channel, const void* data, size_t len));
+ MOCK_METHOD2(SendPacket, int(const void* data, size_t len));
+ MOCK_METHOD2(SendRTCPPacket, int(const void* data, size_t len));
};
} // namespace webrtc
#endif // WEBRTC_TEST_MOCK_TRANSPORT_H_
diff --git a/webrtc/video/rtc_event_log_unittest.cc b/webrtc/video/rtc_event_log_unittest.cc
index 6c1786b..cc8a8d2 100644
--- a/webrtc/video/rtc_event_log_unittest.cc
+++ b/webrtc/video/rtc_event_log_unittest.cc
@@ -295,8 +295,7 @@
RTC_CHECK_GE(packet_size, 16 + 4 * csrcs_count + 4 * kNumExtensions);
Clock* clock = Clock::GetRealTimeClock();
- RTPSender rtp_sender(0, // int32_t id
- false, // bool audio
+ RTPSender rtp_sender(false, // bool audio
clock, // Clock* clock
nullptr, // Transport*
nullptr, // RtpAudioFeedback*
diff --git a/webrtc/video/transport_adapter.cc b/webrtc/video/transport_adapter.cc
index e5c9f61..653e55c 100644
--- a/webrtc/video/transport_adapter.cc
+++ b/webrtc/video/transport_adapter.cc
@@ -20,9 +20,7 @@
RTC_DCHECK(nullptr != transport);
}
-int TransportAdapter::SendPacket(int /*channel*/,
- const void* packet,
- size_t length) {
+int TransportAdapter::SendPacket(const void* packet, size_t length) {
if (enabled_.Value() == 0)
return false;
@@ -31,9 +29,7 @@
return success ? static_cast<int>(length) : -1;
}
-int TransportAdapter::SendRTCPPacket(int /*channel*/,
- const void* packet,
- size_t length) {
+int TransportAdapter::SendRTCPPacket(const void* packet, size_t length) {
if (enabled_.Value() == 0)
return false;
diff --git a/webrtc/video/transport_adapter.h b/webrtc/video/transport_adapter.h
index cd27d7c..c9a7938 100644
--- a/webrtc/video/transport_adapter.h
+++ b/webrtc/video/transport_adapter.h
@@ -21,10 +21,8 @@
public:
explicit TransportAdapter(newapi::Transport* transport);
- int SendPacket(int /*channel*/, const void* packet, size_t length) override;
- int SendRTCPPacket(int /*channel*/,
- const void* packet,
- size_t length) override;
+ int SendPacket(const void* packet, size_t length) override;
+ int SendRTCPPacket(const void* packet, size_t length) override;
void Enable();
void Disable();
diff --git a/webrtc/video/video_receive_stream.cc b/webrtc/video/video_receive_stream.cc
index efa97c7..2c8d8f8 100644
--- a/webrtc/video/video_receive_stream.cc
+++ b/webrtc/video/video_receive_stream.cc
@@ -140,7 +140,7 @@
channel_group_(channel_group),
channel_id_(channel_id) {
RTC_CHECK(channel_group_->CreateReceiveChannel(
- channel_id_, 0, &transport_adapter_, num_cpu_cores));
+ channel_id_, &transport_adapter_, num_cpu_cores));
vie_channel_ = channel_group_->GetChannel(channel_id_);
diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc
index 2ab4eaa..462d252 100644
--- a/webrtc/video/video_send_stream.cc
+++ b/webrtc/video/video_send_stream.cc
@@ -118,8 +118,8 @@
use_config_bitrate_(true),
stats_proxy_(Clock::GetRealTimeClock(), config) {
RTC_DCHECK(!config_.rtp.ssrcs.empty());
- RTC_CHECK(channel_group->CreateSendChannel(
- channel_id_, 0, &transport_adapter_, num_cpu_cores, config_.rtp.ssrcs));
+ RTC_CHECK(channel_group->CreateSendChannel(channel_id_, &transport_adapter_,
+ num_cpu_cores, config_.rtp.ssrcs));
vie_channel_ = channel_group_->GetChannel(channel_id_);
vie_encoder_ = channel_group_->GetEncoder(channel_id_);
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index a70490a..2ff411d 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -328,7 +328,7 @@
// Receive statistics reporting having lost 50% of the packets.
FakeReceiveStatistics lossy_receive_stats(
kSendSsrcs[0], header.sequenceNumber, send_count_ / 2, 127);
- RTCPSender rtcp_sender(0, false, Clock::GetRealTimeClock(),
+ RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(),
&lossy_receive_stats, nullptr);
EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
@@ -410,8 +410,8 @@
uint16_t nack_sequence_number = header.sequenceNumber - 1;
nacked_sequence_number_ = nack_sequence_number;
NullReceiveStatistics null_stats;
- RTCPSender rtcp_sender(
- 0, false, Clock::GetRealTimeClock(), &null_stats, nullptr);
+ RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(), &null_stats,
+ nullptr);
EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
@@ -592,7 +592,7 @@
// Receive statistics reporting having lost 50% of the packets.
FakeReceiveStatistics lossy_receive_stats(
kSendSsrcs[0], header.sequenceNumber, packet_count_ / 2, 127);
- RTCPSender rtcp_sender(0, false, Clock::GetRealTimeClock(),
+ RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(),
&lossy_receive_stats, nullptr);
EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
@@ -823,7 +823,7 @@
EXCLUSIVE_LOCKS_REQUIRED(crit_) {
FakeReceiveStatistics receive_stats(
kSendSsrcs[0], last_sequence_number_, rtp_count_, 0);
- RTCPSender rtcp_sender(0, false, clock_, &receive_stats, nullptr);
+ RTCPSender rtcp_sender(false, clock_, &receive_stats, nullptr);
EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
@@ -882,8 +882,8 @@
observation_complete_->Set();
// Receive statistics reporting having lost 50% of the packets.
FakeReceiveStatistics receive_stats(kSendSsrcs[0], 1, 1, 0);
- RTCPSender rtcp_sender(0, false, Clock::GetRealTimeClock(),
- &receive_stats, nullptr);
+ RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(), &receive_stats,
+ nullptr);
EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
diff --git a/webrtc/video_engine/vie_channel.cc b/webrtc/video_engine/vie_channel.cc
index e941326..b812926 100644
--- a/webrtc/video_engine/vie_channel.cc
+++ b/webrtc/video_engine/vie_channel.cc
@@ -77,9 +77,7 @@
ViEChannel* owner_;
};
-ViEChannel::ViEChannel(int32_t channel_id,
- int32_t engine_id,
- uint32_t number_of_cores,
+ViEChannel::ViEChannel(uint32_t number_of_cores,
Transport* transport,
ProcessThread* module_process_thread,
RtcpIntraFrameObserver* intra_frame_observer,
@@ -91,9 +89,7 @@
PacketRouter* packet_router,
size_t max_rtp_streams,
bool sender)
- : channel_id_(channel_id),
- engine_id_(engine_id),
- number_of_cores_(number_of_cores),
+ : number_of_cores_(number_of_cores),
sender_(sender),
module_process_thread_(module_process_thread),
crit_(CriticalSectionWrapper::CreateCriticalSection()),
@@ -102,7 +98,7 @@
vcm_(VideoCodingModule::Create(Clock::GetRealTimeClock(),
nullptr,
nullptr)),
- vie_receiver_(channel_id, vcm_, remote_bitrate_estimator, this),
+ vie_receiver_(vcm_, remote_bitrate_estimator, this),
vie_sync_(vcm_),
stats_observer_(new ChannelStatsObserver(this)),
receive_stats_callback_(nullptr),
@@ -121,8 +117,7 @@
rtt_sum_ms_(0),
num_rtts_(0),
rtp_rtcp_modules_(
- CreateRtpRtcpModules(ViEModuleId(engine_id_, channel_id_),
- !sender,
+ CreateRtpRtcpModules(!sender,
vie_receiver_.GetReceiveStatistics(),
transport,
sender ? intra_frame_observer_ : nullptr,
@@ -1023,11 +1018,11 @@
int32_t ViEChannel::FrameToRender(VideoFrame& video_frame) { // NOLINT
CriticalSectionScoped cs(crit_.get());
-
if (pre_render_callback_ != NULL)
pre_render_callback_->FrameCallback(&video_frame);
- incoming_video_stream_->RenderFrame(channel_id_, video_frame);
+ // TODO(pbos): Remove stream id argument.
+ incoming_video_stream_->RenderFrame(0xFFFFFFFF, video_frame);
return 0;
}
@@ -1134,7 +1129,6 @@
}
std::vector<RtpRtcp*> ViEChannel::CreateRtpRtcpModules(
- int32_t id,
bool receiver_only,
ReceiveStatistics* receive_statistics,
Transport* outgoing_transport,
@@ -1153,7 +1147,6 @@
RTC_DCHECK_GT(num_modules, 0u);
RtpRtcp::Configuration configuration;
ReceiveStatistics* null_receive_statistics = configuration.receive_statistics;
- configuration.id = id;
configuration.audio = false;
configuration.receiver_only = receiver_only;
configuration.receive_statistics = receive_statistics;
@@ -1231,7 +1224,6 @@
// TODO(pbos): Remove OnInitializeDecoder which is called from the RTP module,
// any decoder resetting should be handled internally within the VCM.
int32_t ViEChannel::OnInitializeDecoder(
- const int32_t id,
const int8_t payload_type,
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const int frequency,
@@ -1244,17 +1236,11 @@
return 0;
}
-void ViEChannel::OnIncomingSSRCChanged(const int32_t id, const uint32_t ssrc) {
- RTC_DCHECK_EQ(channel_id_, ChannelId(id));
+void ViEChannel::OnIncomingSSRCChanged(const uint32_t ssrc) {
rtp_rtcp_modules_[0]->SetRemoteSSRC(ssrc);
}
-void ViEChannel::OnIncomingCSRCChanged(const int32_t id,
- const uint32_t CSRC,
- const bool added) {
- RTC_DCHECK_EQ(channel_id_, ChannelId(id));
- CriticalSectionScoped cs(crit_.get());
-}
+void ViEChannel::OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) {}
void ViEChannel::RegisterSendFrameCountObserver(
FrameCountObserver* observer) {
diff --git a/webrtc/video_engine/vie_channel.h b/webrtc/video_engine/vie_channel.h
index 834c933..fc01592 100644
--- a/webrtc/video_engine/vie_channel.h
+++ b/webrtc/video_engine/vie_channel.h
@@ -65,9 +65,7 @@
friend class ChannelStatsObserver;
friend class ViEChannelProtectionCallback;
- ViEChannel(int32_t channel_id,
- int32_t engine_id,
- uint32_t number_of_cores,
+ ViEChannel(uint32_t number_of_cores,
Transport* transport,
ProcessThread* module_process_thread,
RtcpIntraFrameObserver* intra_frame_observer,
@@ -192,18 +190,13 @@
void RegisterSendBitrateObserver(BitrateStatisticsObserver* observer);
// Implements RtpFeedback.
- virtual int32_t OnInitializeDecoder(
- const int32_t id,
- const int8_t payload_type,
- const char payload_name[RTP_PAYLOAD_NAME_SIZE],
- const int frequency,
- const uint8_t channels,
- const uint32_t rate);
- virtual void OnIncomingSSRCChanged(const int32_t id,
- const uint32_t ssrc);
- virtual void OnIncomingCSRCChanged(const int32_t id,
- const uint32_t CSRC,
- const bool added);
+ int32_t OnInitializeDecoder(const int8_t payload_type,
+ const char payload_name[RTP_PAYLOAD_NAME_SIZE],
+ const int frequency,
+ const uint8_t channels,
+ const uint32_t rate) override;
+ void OnIncomingSSRCChanged(const uint32_t ssrc) override;
+ void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) override;
int32_t SetRemoteSSRCType(const StreamType usage, const uint32_t SSRC);
@@ -296,7 +289,6 @@
private:
static std::vector<RtpRtcp*> CreateRtpRtcpModules(
- int32_t id,
bool receiver_only,
ReceiveStatistics* receive_statistics,
Transport* outgoing_transport,
@@ -410,8 +402,6 @@
GUARDED_BY(critsect_);
} rtcp_packet_type_counter_observer_;
- const int32_t channel_id_;
- const int32_t engine_id_;
const uint32_t number_of_cores_;
const bool sender_;
diff --git a/webrtc/video_engine/vie_channel_group.cc b/webrtc/video_engine/vie_channel_group.cc
index 5c55aaa..42f6c85 100644
--- a/webrtc/video_engine/vie_channel_group.cc
+++ b/webrtc/video_engine/vie_channel_group.cc
@@ -186,7 +186,6 @@
}
bool ChannelGroup::CreateSendChannel(int channel_id,
- int engine_id,
Transport* transport,
int number_of_cores,
const std::vector<uint32_t>& ssrcs) {
@@ -198,7 +197,7 @@
return false;
}
ViEEncoder* encoder = vie_encoder.get();
- if (!CreateChannel(channel_id, engine_id, transport, number_of_cores,
+ if (!CreateChannel(channel_id, transport, number_of_cores,
vie_encoder.release(), ssrcs.size(), true)) {
return false;
}
@@ -214,22 +213,20 @@
}
bool ChannelGroup::CreateReceiveChannel(int channel_id,
- int engine_id,
Transport* transport,
int number_of_cores) {
- return CreateChannel(channel_id, engine_id, transport, number_of_cores,
+ return CreateChannel(channel_id, transport, number_of_cores,
nullptr, 1, false);
}
bool ChannelGroup::CreateChannel(int channel_id,
- int engine_id,
Transport* transport,
int number_of_cores,
ViEEncoder* vie_encoder,
size_t max_rtp_streams,
bool sender) {
rtc::scoped_ptr<ViEChannel> channel(new ViEChannel(
- channel_id, engine_id, number_of_cores, transport, process_thread_,
+ number_of_cores, transport, process_thread_,
encoder_state_feedback_->GetRtcpIntraFrameObserver(),
bitrate_controller_->CreateRtcpBandwidthObserver(), nullptr,
remote_bitrate_estimator_.get(), call_stats_->rtcp_rtt_stats(),
diff --git a/webrtc/video_engine/vie_channel_group.h b/webrtc/video_engine/vie_channel_group.h
index f1faa80..ec17083 100644
--- a/webrtc/video_engine/vie_channel_group.h
+++ b/webrtc/video_engine/vie_channel_group.h
@@ -44,12 +44,10 @@
explicit ChannelGroup(ProcessThread* process_thread);
~ChannelGroup();
bool CreateSendChannel(int channel_id,
- int engine_id,
Transport* transport,
int number_of_cores,
const std::vector<uint32_t>& ssrcs);
bool CreateReceiveChannel(int channel_id,
- int engine_id,
Transport* transport,
int number_of_cores);
void DeleteChannel(int channel_id);
@@ -75,7 +73,6 @@
typedef std::map<int, ViEEncoder*> EncoderMap;
bool CreateChannel(int channel_id,
- int engine_id,
Transport* transport,
int number_of_cores,
ViEEncoder* vie_encoder,
diff --git a/webrtc/video_engine/vie_receiver.cc b/webrtc/video_engine/vie_receiver.cc
index a3ef2cf..f10f287 100644
--- a/webrtc/video_engine/vie_receiver.cc
+++ b/webrtc/video_engine/vie_receiver.cc
@@ -33,8 +33,7 @@
static const int kPacketLogIntervalMs = 10000;
-ViEReceiver::ViEReceiver(const int32_t channel_id,
- VideoCodingModule* module_vcm,
+ViEReceiver::ViEReceiver(VideoCodingModule* module_vcm,
RemoteBitrateEstimator* remote_bitrate_estimator,
RtpFeedback* rtp_feedback)
: receive_cs_(CriticalSectionWrapper::CreateCriticalSection()),
@@ -43,8 +42,7 @@
rtp_payload_registry_(
new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(false))),
rtp_receiver_(
- RtpReceiver::CreateVideoReceiver(channel_id,
- clock_,
+ RtpReceiver::CreateVideoReceiver(clock_,
this,
rtp_feedback,
rtp_payload_registry_.get())),
diff --git a/webrtc/video_engine/vie_receiver.h b/webrtc/video_engine/vie_receiver.h
index 781f14f..c7d4c33 100644
--- a/webrtc/video_engine/vie_receiver.h
+++ b/webrtc/video_engine/vie_receiver.h
@@ -36,7 +36,7 @@
class ViEReceiver : public RtpData {
public:
- ViEReceiver(const int32_t channel_id, VideoCodingModule* module_vcm,
+ ViEReceiver(VideoCodingModule* module_vcm,
RemoteBitrateEstimator* remote_bitrate_estimator,
RtpFeedback* rtp_feedback);
~ViEReceiver();
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index a8fb179..7f315df 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -218,14 +218,10 @@
}
int
-Channel::SendPacket(int channel, const void *data, size_t len)
+Channel::SendPacket(const void *data, size_t len)
{
- channel = VoEChannelId(channel);
- assert(channel == _channelId);
-
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SendPacket(channel=%d, len=%" PRIuS ")", channel,
- len);
+ "Channel::SendPacket(channel=%d, len=%" PRIuS ")", len);
CriticalSectionScoped cs(&_callbackCritSect);
@@ -240,8 +236,7 @@
uint8_t* bufferToSendPtr = (uint8_t*)data;
size_t bufferLength = len;
- int n = _transportPtr->SendPacket(channel, bufferToSendPtr,
- bufferLength);
+ int n = _transportPtr->SendPacket(bufferToSendPtr, bufferLength);
if (n < 0) {
std::string transport_name =
_externalTransport ? "external transport" : "WebRtc sockets";
@@ -255,14 +250,10 @@
}
int
-Channel::SendRTCPPacket(int channel, const void *data, size_t len)
+Channel::SendRTCPPacket(const void *data, size_t len)
{
- channel = VoEChannelId(channel);
- assert(channel == _channelId);
-
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SendRTCPPacket(channel=%d, len=%" PRIuS ")", channel,
- len);
+ "Channel::SendRTCPPacket(len=%" PRIuS ")", len);
CriticalSectionScoped cs(&_callbackCritSect);
if (_transportPtr == NULL)
@@ -277,9 +268,7 @@
uint8_t* bufferToSendPtr = (uint8_t*)data;
size_t bufferLength = len;
- int n = _transportPtr->SendRTCPPacket(channel,
- bufferToSendPtr,
- bufferLength);
+ int n = _transportPtr->SendRTCPPacket(bufferToSendPtr, bufferLength);
if (n < 0) {
std::string transport_name =
_externalTransport ? "external transport" : "WebRtc sockets";
@@ -292,15 +281,12 @@
return n;
}
-void
-Channel::OnPlayTelephoneEvent(int32_t id,
- uint8_t event,
- uint16_t lengthMs,
- uint8_t volume)
-{
+void Channel::OnPlayTelephoneEvent(uint8_t event,
+ uint16_t lengthMs,
+ uint8_t volume) {
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::OnPlayTelephoneEvent(id=%d, event=%u, lengthMs=%u,"
- " volume=%u)", id, event, lengthMs, volume);
+ "Channel::OnPlayTelephoneEvent(event=%u, lengthMs=%u,"
+ " volume=%u)", event, lengthMs, volume);
if (!_playOutbandDtmfEvent || (event > 15))
{
@@ -317,40 +303,31 @@
}
void
-Channel::OnIncomingSSRCChanged(int32_t id, uint32_t ssrc)
+Channel::OnIncomingSSRCChanged(uint32_t ssrc)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::OnIncomingSSRCChanged(id=%d, SSRC=%d)",
- id, ssrc);
+ "Channel::OnIncomingSSRCChanged(SSRC=%d)", ssrc);
// Update ssrc so that NTP for AV sync can be updated.
_rtpRtcpModule->SetRemoteSSRC(ssrc);
}
-void Channel::OnIncomingCSRCChanged(int32_t id,
- uint32_t CSRC,
- bool added)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::OnIncomingCSRCChanged(id=%d, CSRC=%d, added=%d)",
- id, CSRC, added);
+void Channel::OnIncomingCSRCChanged(uint32_t CSRC, bool added) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+ "Channel::OnIncomingCSRCChanged(CSRC=%d, added=%d)", CSRC,
+ added);
}
-int32_t
-Channel::OnInitializeDecoder(
- int32_t id,
+int32_t Channel::OnInitializeDecoder(
int8_t payloadType,
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
int frequency,
uint8_t channels,
- uint32_t rate)
-{
+ uint32_t rate) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::OnInitializeDecoder(id=%d, payloadType=%d, "
+ "Channel::OnInitializeDecoder(payloadType=%d, "
"payloadName=%s, frequency=%u, channels=%u, rate=%u)",
- id, payloadType, payloadName, frequency, channels, rate);
-
- assert(VoEChannelId(id) == _channelId);
+ payloadType, payloadName, frequency, channels, rate);
CodecInst receiveCodec = {0};
CodecInst dummyCodec = {0};
@@ -732,8 +709,7 @@
rtp_receive_statistics_(
ReceiveStatistics::Create(Clock::GetRealTimeClock())),
rtp_receiver_(
- RtpReceiver::CreateAudioReceiver(VoEModuleId(instanceId, channelId),
- Clock::GetRealTimeClock(),
+ RtpReceiver::CreateAudioReceiver(Clock::GetRealTimeClock(),
this,
this,
this,
@@ -823,7 +799,6 @@
_outputAudioLevel.Clear();
RtpRtcp::Configuration configuration;
- configuration.id = VoEModuleId(instanceId, channelId);
configuration.audio = true;
configuration.outgoing_transport = this;
configuration.audio_messages = this;
@@ -3896,11 +3871,11 @@
}
if (!RTCP)
{
- return _transportPtr->SendPacket(_channelId, data, len);
+ return _transportPtr->SendPacket(data, len);
}
else
{
- return _transportPtr->SendRTCPPacket(_channelId, data, len);
+ return _transportPtr->SendRTCPPacket(data, len);
}
}
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 66eaff1..d9e4575 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -364,24 +364,22 @@
size_t packet_length) override;
// From RtpFeedback in the RTP/RTCP module
- int32_t OnInitializeDecoder(int32_t id,
- int8_t payloadType,
+ int32_t OnInitializeDecoder(int8_t payloadType,
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
int frequency,
uint8_t channels,
uint32_t rate) override;
- void OnIncomingSSRCChanged(int32_t id, uint32_t ssrc) override;
- void OnIncomingCSRCChanged(int32_t id, uint32_t CSRC, bool added) override;
+ void OnIncomingSSRCChanged(uint32_t ssrc) override;
+ void OnIncomingCSRCChanged(uint32_t CSRC, bool added) override;
// From RtpAudioFeedback in the RTP/RTCP module
- void OnPlayTelephoneEvent(int32_t id,
- uint8_t event,
+ void OnPlayTelephoneEvent(uint8_t event,
uint16_t lengthMs,
uint8_t volume) override;
// From Transport (called by the RTP/RTCP module)
- int SendPacket(int /*channel*/, const void* data, size_t len) override;
- int SendRTCPPacket(int /*channel*/, const void* data, size_t len) override;
+ int SendPacket(const void* data, size_t len) override;
+ int SendRTCPPacket(const void* data, size_t len) override;
// From MixerParticipant
int32_t GetAudioFrame(int32_t id, AudioFrame* audioFrame) override;
diff --git a/webrtc/voice_engine/mock/mock_transport.h b/webrtc/voice_engine/mock/mock_transport.h
index 3f80a16..c356f44 100644
--- a/webrtc/voice_engine/mock/mock_transport.h
+++ b/webrtc/voice_engine/mock/mock_transport.h
@@ -18,8 +18,8 @@
class MockTransport : public Transport {
public:
- MOCK_METHOD3(SendPacket, int(int channel, const void* data, size_t len));
- MOCK_METHOD3(SendRTCPPacket, int(int channel, const void* data, size_t len));
+ MOCK_METHOD2(SendPacket, int(const void* data, size_t len));
+ MOCK_METHOD2(SendRTCPPacket, int(const void* data, size_t len));
};
} // namespace webrtc
diff --git a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
index ef245e9..54a6b8e 100644
--- a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
+++ b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
@@ -108,14 +108,13 @@
EXPECT_TRUE(webrtc::VoiceEngine::Delete(local_voe_));
}
-int ConferenceTransport::SendPacket(int channel, const void* data, size_t len) {
- StorePacket(Packet::Rtp, channel, data, len);
+int ConferenceTransport::SendPacket(const void* data, size_t len) {
+ StorePacket(Packet::Rtp, data, len);
return static_cast<int>(len);
}
-int ConferenceTransport::SendRTCPPacket(int channel, const void* data,
- size_t len) {
- StorePacket(Packet::Rtcp, channel, data, len);
+int ConferenceTransport::SendRTCPPacket(const void* data, size_t len) {
+ StorePacket(Packet::Rtcp, data, len);
return static_cast<int>(len);
}
@@ -129,11 +128,12 @@
return -1;
}
-void ConferenceTransport::StorePacket(Packet::Type type, int channel,
- const void* data, size_t len) {
+void ConferenceTransport::StorePacket(Packet::Type type,
+ const void* data,
+ size_t len) {
{
webrtc::CriticalSectionScoped lock(pq_crit_.get());
- packet_queue_.push_back(Packet(type, channel, data, len, rtc::Time()));
+ packet_queue_.push_back(Packet(type, data, len, rtc::Time()));
}
packet_event_->Set();
}
diff --git a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
index 10bf411..e7afccb 100644
--- a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
+++ b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
@@ -98,24 +98,20 @@
bool GetReceiverStatistics(unsigned int id, webrtc::CallStatistics* stats);
// Inherit from class webrtc::Transport.
- int SendPacket(int channel, const void *data, size_t len) override;
- int SendRTCPPacket(int channel, const void *data, size_t len) override;
+ int SendPacket(const void *data, size_t len) override;
+ int SendRTCPPacket(const void *data, size_t len) override;
private:
struct Packet {
enum Type { Rtp, Rtcp, } type_;
Packet() : len_(0) {}
- Packet(Type type, int channel, const void* data, size_t len, uint32 time_ms)
- : type_(type),
- channel_(channel),
- len_(len),
- send_time_ms_(time_ms) {
+ Packet(Type type, const void* data, size_t len, uint32 time_ms)
+ : type_(type), len_(len), send_time_ms_(time_ms) {
EXPECT_LE(len_, kMaxPacketSizeByte);
memcpy(data_, data, len_);
}
- int channel_;
uint8_t data_[kMaxPacketSizeByte];
size_t len_;
uint32 send_time_ms_;
@@ -126,8 +122,7 @@
}
int GetReceiverChannelForSsrc(unsigned int sender_ssrc) const;
- void StorePacket(Packet::Type type, int channel, const void* data,
- size_t len);
+ void StorePacket(Packet::Type type, const void* data, size_t len);
void SendPacket(const Packet& packet);
bool DispatchPackets();
diff --git a/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h b/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h
index cee5a58..11397c1 100644
--- a/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h
+++ b/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h
@@ -15,6 +15,7 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/system_wrappers/interface/atomic32.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
@@ -26,24 +27,27 @@
class LoopBackTransport : public webrtc::Transport {
public:
- LoopBackTransport(webrtc::VoENetwork* voe_network)
+ LoopBackTransport(webrtc::VoENetwork* voe_network, int channel)
: crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()),
packet_event_(webrtc::EventWrapper::Create()),
- thread_(webrtc::ThreadWrapper::CreateThread(
- NetworkProcess, this, "LoopBackTransport")),
- voe_network_(voe_network), transmitted_packets_(0) {
+ thread_(webrtc::ThreadWrapper::CreateThread(NetworkProcess,
+ this,
+ "LoopBackTransport")),
+ channel_(channel),
+ voe_network_(voe_network),
+ transmitted_packets_(0) {
thread_->Start();
}
~LoopBackTransport() { thread_->Stop(); }
- int SendPacket(int channel, const void* data, size_t len) override {
- StorePacket(Packet::Rtp, channel, data, len);
+ int SendPacket(const void* data, size_t len) override {
+ StorePacket(Packet::Rtp, data, len);
return static_cast<int>(len);
}
- int SendRTCPPacket(int channel, const void* data, size_t len) override {
- StorePacket(Packet::Rtcp, channel, data, len);
+ int SendRTCPPacket(const void* data, size_t len) override {
+ StorePacket(Packet::Rtcp, data, len);
return static_cast<int>(len);
}
@@ -57,28 +61,32 @@
}
}
+ void AddChannel(uint32_t ssrc, int channel) {
+ webrtc::CriticalSectionScoped lock(crit_.get());
+ channels_[ssrc] = channel;
+ }
+
private:
struct Packet {
enum Type { Rtp, Rtcp, } type;
Packet() : len(0) {}
- Packet(Type type, int channel, const void* data, size_t len)
- : type(type), channel(channel), len(len) {
+ Packet(Type type, const void* data, size_t len)
+ : type(type), len(len) {
assert(len <= 1500);
memcpy(this->data, data, len);
}
- int channel;
uint8_t data[1500];
size_t len;
};
- void StorePacket(Packet::Type type, int channel,
+ void StorePacket(Packet::Type type,
const void* data,
size_t len) {
{
webrtc::CriticalSectionScoped lock(crit_.get());
- packet_queue_.push_back(Packet(type, channel, data, len));
+ packet_queue_.push_back(Packet(type, data, len));
}
packet_event_->Set();
}
@@ -100,21 +108,34 @@
while (true) {
Packet p;
+ int channel = channel_;
{
webrtc::CriticalSectionScoped lock(crit_.get());
if (packet_queue_.empty())
break;
p = packet_queue_.front();
packet_queue_.pop_front();
+
+ if (p.type == Packet::Rtp) {
+ uint32_t ssrc =
+ webrtc::ByteReader<uint32_t>::ReadBigEndian(&p.data[8]);
+ if (channels_[ssrc] != 0)
+ channel = channels_[ssrc];
+ }
+ // TODO(pbos): Add RTCP SSRC muxing/demuxing if anything requires it.
}
+ // Minimum RTP header size.
+ if (p.len < 12)
+ continue;
+
switch (p.type) {
case Packet::Rtp:
- voe_network_->ReceivedRTPPacket(p.channel, p.data, p.len,
+ voe_network_->ReceivedRTPPacket(channel, p.data, p.len,
webrtc::PacketTime());
break;
case Packet::Rtcp:
- voe_network_->ReceivedRTCPPacket(p.channel, p.data, p.len);
+ voe_network_->ReceivedRTCPPacket(channel, p.data, p.len);
break;
}
++transmitted_packets_;
@@ -126,6 +147,8 @@
const rtc::scoped_ptr<webrtc::EventWrapper> packet_event_;
const rtc::scoped_ptr<webrtc::ThreadWrapper> thread_;
std::deque<Packet> packet_queue_ GUARDED_BY(crit_.get());
+ const int channel_;
+ std::map<uint32_t, int> channels_ GUARDED_BY(crit_.get());
webrtc::VoENetwork* const voe_network_;
webrtc::Atomic32 transmitted_packets_;
};
diff --git a/webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.cc b/webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.cc
index 488f448..abcb2a6 100644
--- a/webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.cc
+++ b/webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.cc
@@ -61,7 +61,7 @@
}
void BeforeStreamingFixture::SetUpLocalPlayback() {
- transport_ = new LoopBackTransport(voe_network_);
+ transport_ = new LoopBackTransport(voe_network_, channel_);
EXPECT_EQ(0, voe_network_->RegisterExternalTransport(channel_, *transport_));
webrtc::CodecInst codec;
diff --git a/webrtc/voice_engine/test/auto_test/standard/mixing_test.cc b/webrtc/voice_engine/test/auto_test/standard/mixing_test.cc
index 2a5732b..3c61956 100644
--- a/webrtc/voice_engine/test/auto_test/standard/mixing_test.cc
+++ b/webrtc/voice_engine/test/auto_test/standard/mixing_test.cc
@@ -35,7 +35,7 @@
: output_filename_(test::OutputPath() + "mixing_test_output.pcm") {
}
void SetUp() {
- transport_ = new LoopBackTransport(voe_network_);
+ transport_ = new LoopBackTransport(voe_network_, 0);
}
void TearDown() {
delete transport_;
@@ -182,6 +182,9 @@
void StartRemoteStream(int stream, const CodecInst& codec_inst, int port) {
EXPECT_EQ(0, voe_codec_->SetRecPayloadType(stream, codec_inst));
EXPECT_EQ(0, voe_network_->RegisterExternalTransport(stream, *transport_));
+ EXPECT_EQ(0, voe_rtp_rtcp_->SetLocalSSRC(
+ stream, static_cast<unsigned int>(stream)));
+ transport_->AddChannel(stream, stream);
EXPECT_EQ(0, voe_base_->StartReceive(stream));
EXPECT_EQ(0, voe_base_->StartPlayout(stream));
EXPECT_EQ(0, voe_codec_->SetSendCodec(stream, codec_inst));
diff --git a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
index 7344338..4f92e06 100644
--- a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
+++ b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
@@ -28,7 +28,7 @@
audio_level_id_(-1),
absolute_sender_time_id_(-1) {}
- int SendPacket(int channel, const void* data, size_t len) override {
+ int SendPacket(const void* data, size_t len) override {
webrtc::RTPHeader header;
if (parser_->Parse(reinterpret_cast<const uint8_t*>(data), len, &header)) {
bool ok = true;
@@ -51,7 +51,7 @@
return static_cast<int>(len);
}
- int SendRTCPPacket(int channel, const void* data, size_t len) override {
+ int SendRTCPPacket(const void* data, size_t len) override {
return static_cast<int>(len);
}
diff --git a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
index 236232b..9574646 100644
--- a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
+++ b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
@@ -63,7 +63,7 @@
second_channel_ = voe_base_->CreateChannel();
EXPECT_GE(second_channel_, 0);
- transport_ = new LoopBackTransport(voe_network_);
+ transport_ = new LoopBackTransport(voe_network_, second_channel_);
EXPECT_EQ(0, voe_network_->RegisterExternalTransport(second_channel_,
*transport_));
