Move ownership of RTPSenderAudio to ChannelSend.
This change takes out responsibility for packetization from the
RtpRtcp class, and deletes the method RtpRtcp::SendOutgoingData.
Video packetization was similarly moved in cl
https://webrtc-review.googlesource.com/c/src/+/123187
Bug: webrtc:7135
Change-Id: I0953125a5ca22a2ce51761b83693e0bb8ea74cd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27000}diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index d3ec157..e12e245 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -796,8 +796,7 @@
void AudioSendStream::RegisterCngPayloadType(int payload_type,
int clockrate_hz) {
- rtp_rtcp_module_->RegisterAudioSendPayload(payload_type, "CN", clockrate_hz,
- 1, 0);
+ channel_send_->RegisterCngPayloadType(payload_type, clockrate_hz);
}
} // namespace internal
} // namespace webrtc
diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 1d4d5b7..df5ee30 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -450,6 +450,7 @@
stolen_encoder = std::move(*encoder);
return true;
}));
+ EXPECT_CALL(*helper.channel_send(), RegisterCngPayloadType(105, 8000));
auto send_stream = helper.CreateAudioSendStream();
@@ -501,6 +502,8 @@
EXPECT_CALL(*helper.channel_send(), SetEncoderForMock(_, _))
.WillOnce(Return());
+ EXPECT_CALL(*helper.channel_send(), RegisterCngPayloadType(105, 8000));
+
helper.config().send_codec_spec =
AudioSendStream::Config::SendCodecSpec(9, kG722Format);
helper.config().send_codec_spec->cng_payload_type = 105;
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 705827d..196911a 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -129,6 +129,8 @@
// Used by AudioSendStream.
RtpRtcp* GetRtpRtcp() const override;
+ void RegisterCngPayloadType(int payload_type, int payload_frequency) override;
+
// DTMF.
bool SendTelephoneEventOutband(int event, int duration_ms) override;
void SetSendTelephoneEventPayloadType(int payload_type,
@@ -236,6 +238,7 @@
RtcEventLog* const event_log_;
std::unique_ptr<RtpRtcp> _rtpRtcpModule;
+ std::unique_ptr<RTPSenderAudio> rtp_sender_audio_;
std::unique_ptr<AudioCodingModule> audio_coding_;
uint32_t _timeStamp RTC_GUARDED_BY(encoder_queue_);
@@ -520,10 +523,10 @@
const RTPFragmentationHeader* fragmentation) {
RTC_DCHECK_RUN_ON(encoder_queue_);
if (_includeAudioLevelIndication) {
- // Store current audio level in the RTP/RTCP module.
+ // Store current audio level in the RTP sender.
// The level will be used in combination with voice-activity state
// (frameType) to add an RTP header extension
- _rtpRtcpModule->SetAudioLevel(rms_level_.Average());
+ rtp_sender_audio_->SetAudioLevel(rms_level_.Average());
}
// E2EE Custom Audio Frame Encryption (This is optional).
@@ -559,14 +562,24 @@
// Push data from ACM to RTP/RTCP-module to deliver audio frame for
// packetization.
+ if (!_rtpRtcpModule->OnSendingRtpFrame(timeStamp,
+ // Leaving the time when this frame was
+ // received from the capture device as
+ // undefined for voice for now.
+ -1, payloadType,
+ /*force_sender_report=*/false)) {
+ return false;
+ }
+
+ // RTCPSender has it's own copy of the timestamp offset, added in
+ // RTCPSender::BuildSR, hence we must not add the in the offset for the above
+ // call.
+ // TODO(nisse): Delete RTCPSender:timestamp_offset_, and see if we can confine
+ // knowledge of the offset to a single place.
+ const uint32_t rtp_timestamp = timeStamp + _rtpRtcpModule->StartTimestamp();
// This call will trigger Transport::SendPacket() from the RTP/RTCP module.
- if (!_rtpRtcpModule->SendOutgoingData((FrameType&)frameType, payloadType,
- timeStamp,
- // Leaving the time when this frame was
- // received from the capture device as
- // undefined for voice for now.
- -1, payload.data(), payload.size(),
- fragmentation, nullptr, nullptr)) {
+ if (!rtp_sender_audio_->SendAudio(frameType, payloadType, rtp_timestamp,
+ payload.data(), payload.size())) {
RTC_DLOG(LS_ERROR)
<< "ChannelSend::SendData() failed to send data to RTP/RTCP module";
return -1;
@@ -687,6 +700,7 @@
configuration.clock = clock;
configuration.audio = true;
+ configuration.clock = Clock::GetRealTimeClock();
configuration.outgoing_transport = rtp_transport;
configuration.paced_sender = rtp_packet_sender_proxy_.get();
@@ -703,6 +717,9 @@
_rtpRtcpModule = RtpRtcp::Create(configuration);
_rtpRtcpModule->SetSendingMediaStatus(false);
+ rtp_sender_audio_ = absl::make_unique<RTPSenderAudio>(
+ configuration.clock, _rtpRtcpModule->RtpSender());
+
// We want to invoke the 'TargetRateObserver' and |OnOverheadChanged|
// callbacks after the audio_coding_ is fully initialized.
if (media_transport_) {
@@ -797,11 +814,11 @@
// The RTP/RTCP module needs to know the RTP timestamp rate (i.e. clockrate)
// as well as some other things, so we collect this info and send it along.
- _rtpRtcpModule->RegisterAudioSendPayload(payload_type,
- "audio",
- encoder->RtpTimestampRateHz(),
- encoder->NumChannels(),
- 0);
+ _rtpRtcpModule->RegisterSendPayloadFrequency(payload_type,
+ encoder->RtpTimestampRateHz());
+ rtp_sender_audio_->RegisterAudioPayload("audio", payload_type,
+ encoder->RtpTimestampRateHz(),
+ encoder->NumChannels(), 0);
if (media_transport_) {
rtc::CritScope cs(&media_transport_lock_);
@@ -926,21 +943,29 @@
if (!sending_) {
return false;
}
- if (_rtpRtcpModule->SendTelephoneEventOutband(
+ if (rtp_sender_audio_->SendTelephoneEvent(
event, duration_ms, kTelephoneEventAttenuationdB) != 0) {
- RTC_DLOG(LS_ERROR) << "SendTelephoneEventOutband() failed to send event";
+ RTC_DLOG(LS_ERROR) << "SendTelephoneEvent() failed to send event";
return false;
}
return true;
}
+void ChannelSend::RegisterCngPayloadType(int payload_type,
+ int payload_frequency) {
+ _rtpRtcpModule->RegisterSendPayloadFrequency(payload_type, payload_frequency);
+ rtp_sender_audio_->RegisterAudioPayload("CN", payload_type, payload_frequency,
+ 1, 0);
+}
+
void ChannelSend::SetSendTelephoneEventPayloadType(int payload_type,
int payload_frequency) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RTC_DCHECK_LE(0, payload_type);
RTC_DCHECK_GE(127, payload_type);
- _rtpRtcpModule->RegisterAudioSendPayload(payload_type, "telephone-event",
- payload_frequency, 0, 0);
+ _rtpRtcpModule->RegisterSendPayloadFrequency(payload_type, payload_frequency);
+ rtp_sender_audio_->RegisterAudioPayload("telephone-event", payload_type,
+ payload_frequency, 0, 0);
}
void ChannelSend::SetLocalSSRC(uint32_t ssrc) {
diff --git a/audio/channel_send.h b/audio/channel_send.h
index 7149bbd..0957035 100644
--- a/audio/channel_send.h
+++ b/audio/channel_send.h
@@ -20,6 +20,7 @@
#include "api/crypto/crypto_options.h"
#include "api/media_transport_interface.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "modules/rtp_rtcp/source/rtp_sender_audio.h"
#include "rtc_base/function_view.h"
#include "rtc_base/task_queue.h"
@@ -81,6 +82,8 @@
virtual void ResetSenderCongestionControlObjects() = 0;
virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const = 0;
virtual ANAStats GetANAStatistics() const = 0;
+ virtual void RegisterCngPayloadType(int payload_type,
+ int payload_frequency) = 0;
virtual void SetSendTelephoneEventPayloadType(int payload_type,
int payload_frequency) = 0;
virtual bool SendTelephoneEventOutband(int event, int duration_ms) = 0;
diff --git a/audio/mock_voe_channel_proxy.h b/audio/mock_voe_channel_proxy.h
index 1b2f96e..cf2fe88 100644
--- a/audio/mock_voe_channel_proxy.h
+++ b/audio/mock_voe_channel_proxy.h
@@ -95,6 +95,8 @@
MOCK_CONST_METHOD0(GetRTCPStatistics, CallSendStatistics());
MOCK_CONST_METHOD0(GetRemoteRTCPReportBlocks, std::vector<ReportBlock>());
MOCK_CONST_METHOD0(GetANAStatistics, ANAStats());
+ MOCK_METHOD2(RegisterCngPayloadType,
+ void(int payload_type, int payload_frequency));
MOCK_METHOD2(SetSendTelephoneEventPayloadType,
void(int payload_type, int payload_frequency));
MOCK_METHOD2(SendTelephoneEventOutband, bool(int event, int duration_ms));
diff --git a/modules/rtp_rtcp/include/rtp_rtcp.h b/modules/rtp_rtcp/include/rtp_rtcp.h
index 8e1f98b..34ccfeb 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp.h
@@ -146,11 +146,6 @@
// FEC/ULP/RED overhead (when FEC is enabled).
virtual size_t MaxRtpPacketSize() const = 0;
- virtual void RegisterAudioSendPayload(int payload_type,
- absl::string_view payload_name,
- int frequency,
- int channels,
- int rate) = 0;
virtual void RegisterSendPayloadFrequency(int payload_type,
int payload_frequency) = 0;
@@ -257,27 +252,6 @@
virtual RTPSender* RtpSender() = 0;
virtual const RTPSender* RtpSender() const = 0;
- // Used by the codec module to deliver a video or audio frame for
- // packetization.
- // |frame_type| - type of frame to send
- // |payload_type| - payload type of frame to send
- // |timestamp| - timestamp of frame to send
- // |payload_data| - payload buffer of frame to send
- // |payload_size| - size of payload buffer to send
- // |fragmentation| - fragmentation offset data for fragmented frames such
- // as layers or RED
- // |transport_frame_id_out| - set to RTP timestamp.
- // Returns true on success.
- virtual bool SendOutgoingData(FrameType frame_type,
- int8_t payload_type,
- uint32_t timestamp,
- int64_t capture_time_ms,
- const uint8_t* payload_data,
- size_t payload_size,
- const RTPFragmentationHeader* fragmentation,
- const RTPVideoHeader* rtp_video_header,
- uint32_t* transport_frame_id_out) = 0;
-
// Record that a frame is about to be sent. Returns true on success, and false
// if the module isn't ready to send.
virtual bool OnSendingRtpFrame(uint32_t timestamp,
@@ -432,23 +406,6 @@
const VideoBitrateAllocation& bitrate) = 0;
// **************************************************************************
- // Audio
- // **************************************************************************
-
- // Sends a TelephoneEvent tone using RFC 2833 (4733).
- // Returns -1 on failure else 0.
- virtual int32_t SendTelephoneEventOutband(uint8_t key,
- uint16_t time_ms,
- uint8_t level) = 0;
-
- // Store the audio level in dBov for header-extension-for-audio-level-
- // indication.
- // This API shall be called before transmision of an RTP packet to ensure
- // that the |level| part of the extended RTP header is updated.
- // return -1 on failure else 0.
- virtual int32_t SetAudioLevel(uint8_t level_dbov) = 0;
-
- // **************************************************************************
// Video
// **************************************************************************
diff --git a/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index 7d4f0f1..668d527 100644
--- a/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -37,12 +37,6 @@
MOCK_METHOD1(SetRemoteSSRC, void(uint32_t ssrc));
MOCK_METHOD1(SetMaxRtpPacketSize, void(size_t size));
MOCK_CONST_METHOD0(MaxRtpPacketSize, size_t());
- MOCK_METHOD5(RegisterAudioSendPayload,
- void(int payload_type,
- absl::string_view payload_name,
- int frequency,
- int channels,
- int rate));
MOCK_METHOD2(RegisterSendPayloadFrequency,
void(int payload_type, int frequency));
MOCK_METHOD1(DeRegisterSendPayload, int32_t(int8_t payload_type));
@@ -87,16 +81,6 @@
uint32_t* nack_rate));
MOCK_CONST_METHOD1(EstimatedReceiveBandwidth,
int(uint32_t* available_bandwidth));
- MOCK_METHOD9(SendOutgoingData,
- bool(FrameType frame_type,
- int8_t payload_type,
- uint32_t timestamp,
- int64_t capture_time_ms,
- const uint8_t* payload_data,
- size_t payload_size,
- const RTPFragmentationHeader* fragmentation,
- const RTPVideoHeader* rtp_video_header,
- uint32_t* frame_id_out));
MOCK_METHOD4(OnSendingRtpFrame, bool(uint32_t, int64_t, int, bool));
MOCK_METHOD5(TimeToSendPacket,
bool(uint32_t ssrc,
@@ -162,9 +146,6 @@
MOCK_METHOD1(RegisterRtcpStatisticsCallback, void(RtcpStatisticsCallback*));
MOCK_METHOD0(GetRtcpStatisticsCallback, RtcpStatisticsCallback*());
MOCK_METHOD1(SendFeedbackPacket, bool(const rtcp::TransportFeedback& packet));
- MOCK_METHOD3(SendTelephoneEventOutband,
- int32_t(uint8_t key, uint16_t time_ms, uint8_t level));
- MOCK_METHOD1(SetAudioLevel, int32_t(uint8_t level_dbov));
MOCK_METHOD1(SetTargetSendBitrate, void(uint32_t bitrate_bps));
MOCK_METHOD1(SetKeyFrameRequestMethod, int32_t(KeyFrameRequestMethod method));
MOCK_METHOD0(RequestKeyFrame, int32_t());
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 4e06002..6d1c9e2 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -116,9 +116,7 @@
configuration.extmap_allow_mixed,
configuration.field_trials ? *configuration.field_trials
: default_trials));
- if (configuration.audio) {
- audio_ = absl::make_unique<RTPSenderAudio>(clock_, rtp_sender_.get());
- }
+
// Make sure rtcp sender use same timestamp offset as rtp sender.
rtcp_sender_.SetTimestampOffset(rtp_sender_->TimestampOffset());
@@ -270,17 +268,6 @@
rtcp_receiver_.IncomingPacket(rtcp_packet, length);
}
-void ModuleRtpRtcpImpl::RegisterAudioSendPayload(int payload_type,
- absl::string_view payload_name,
- int frequency,
- int channels,
- int rate) {
- RTC_DCHECK(audio_);
- rtcp_sender_.SetRtpClockRate(payload_type, frequency);
- RTC_CHECK_EQ(0, audio_->RegisterAudioPayload(payload_name, payload_type,
- frequency, channels, rate));
-}
-
void ModuleRtpRtcpImpl::RegisterSendPayloadFrequency(int payload_type,
int payload_frequency) {
rtcp_sender_.SetRtpClockRate(payload_type, payload_frequency);
@@ -425,30 +412,6 @@
rtp_sender_->SetAsPartOfAllocation(part_of_allocation);
}
-bool ModuleRtpRtcpImpl::SendOutgoingData(
- FrameType frame_type,
- int8_t payload_type,
- uint32_t time_stamp,
- int64_t capture_time_ms,
- const uint8_t* payload_data,
- size_t payload_size,
- const RTPFragmentationHeader* fragmentation,
- const RTPVideoHeader* rtp_video_header,
- uint32_t* transport_frame_id_out) {
- OnSendingRtpFrame(time_stamp, capture_time_ms, payload_type,
- kVideoFrameKey == frame_type);
-
- const uint32_t rtp_timestamp = time_stamp + rtp_sender_->TimestampOffset();
- if (transport_frame_id_out)
- *transport_frame_id_out = rtp_timestamp;
-
- RTC_DCHECK(audio_);
- RTC_DCHECK(fragmentation == nullptr);
-
- return audio_->SendAudio(frame_type, payload_type, rtp_timestamp,
- payload_data, payload_size);
-}
-
bool ModuleRtpRtcpImpl::OnSendingRtpFrame(uint32_t timestamp,
int64_t capture_time_ms,
int payload_type,
@@ -787,17 +750,6 @@
return rtcp_sender_.SendFeedbackPacket(packet);
}
-// Send a TelephoneEvent tone using RFC 2833 (4733).
-int32_t ModuleRtpRtcpImpl::SendTelephoneEventOutband(const uint8_t key,
- const uint16_t time_ms,
- const uint8_t level) {
- return audio_ ? audio_->SendTelephoneEvent(key, time_ms, level) : -1;
-}
-
-int32_t ModuleRtpRtcpImpl::SetAudioLevel(const uint8_t level_d_bov) {
- return audio_ ? audio_->SetAudioLevel(level_d_bov) : -1;
-}
-
int32_t ModuleRtpRtcpImpl::SetKeyFrameRequestMethod(
const KeyFrameRequestMethod method) {
key_frame_req_method_ = method;
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 20eb179..4b1c927 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -32,8 +32,6 @@
#include "modules/rtp_rtcp/source/rtcp_receiver.h"
#include "modules/rtp_rtcp/source/rtcp_sender.h"
#include "modules/rtp_rtcp/source/rtp_sender.h"
-#include "modules/rtp_rtcp/source/rtp_sender_audio.h"
-#include "modules/rtp_rtcp/source/rtp_sender_video.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/gtest_prod_util.h"
@@ -64,12 +62,6 @@
void SetRemoteSSRC(uint32_t ssrc) override;
// Sender part.
-
- void RegisterAudioSendPayload(int payload_type,
- absl::string_view payload_name,
- int frequency,
- int channels,
- int rate) override;
void RegisterSendPayloadFrequency(int payload_type,
int payload_frequency) override;
@@ -137,18 +129,6 @@
void SetAsPartOfAllocation(bool part_of_allocation) override;
- // Used by the codec module to deliver a video or audio frame for
- // packetization.
- bool SendOutgoingData(FrameType frame_type,
- int8_t payload_type,
- uint32_t time_stamp,
- int64_t capture_time_ms,
- const uint8_t* payload_data,
- size_t payload_size,
- const RTPFragmentationHeader* fragmentation,
- const RTPVideoHeader* rtp_video_header,
- uint32_t* transport_frame_id_out) override;
-
bool OnSendingRtpFrame(uint32_t timestamp,
int64_t capture_time_ms,
int payload_type,
@@ -270,17 +250,6 @@
bool RtcpXrRrtrStatus() const override;
- // Audio part.
-
- // Send a TelephoneEvent tone using RFC 2833 (4733).
- int32_t SendTelephoneEventOutband(uint8_t key,
- uint16_t time_ms,
- uint8_t level) override;
-
- // Store the audio level in d_bov for header-extension-for-audio-level-
- // indication.
- int32_t SetAudioLevel(uint8_t level_d_bov) override;
-
// Video part.
// Set method for requesting a new key frame.
@@ -346,7 +315,6 @@
bool TimeToSendFullNackList(int64_t now) const;
std::unique_ptr<RTPSender> rtp_sender_;
- std::unique_ptr<RTPSenderAudio> audio_;
RTCPSender rtcp_sender_;
RTCPReceiver rtcp_receiver_;
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
index da541a5..98067b2 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
@@ -22,6 +22,7 @@
#include "modules/rtp_rtcp/source/rtcp_packet/nack.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_impl.h"
+#include "modules/rtp_rtcp/source/rtp_sender_video.h"
#include "rtc_base/rate_limiter.h"
#include "test/gmock.h"
#include "test/gtest.h"
diff --git a/modules/rtp_rtcp/source/rtp_sender_audio.cc b/modules/rtp_rtcp/source/rtp_sender_audio.cc
index 56d0884..c049530 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -47,7 +47,9 @@
} // namespace
RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtp_sender)
- : clock_(clock), rtp_sender_(rtp_sender) {}
+ : clock_(clock), rtp_sender_(rtp_sender) {
+ RTC_DCHECK(clock_);
+}
RTPSenderAudio::~RTPSenderAudio() {}