Move ownership of RTPSenderVideo and RTPSenderAudio one level up
From RTPSender to RtpRtcpImpl. Makes RTPSender operate on packets
only, not frames.
Bug: webrtc:7135
Change-Id: Ia9a11456404c3b322d873d4f8fb828742296b26d
Reviewed-on: https://webrtc-review.googlesource.com/c/120044
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26586}diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 4207f7b..3589d6a 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -17,6 +17,7 @@
#include <string>
#include <utility>
+#include "absl/memory/memory.h"
#include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "rtc_base/checks.h"
@@ -83,7 +84,6 @@
: kDefaultVideoReportInterval),
this),
clock_(configuration.clock),
- audio_(configuration.audio),
keepalive_config_(configuration.keepalive_config),
last_bitrate_process_time_(clock_->TimeInMilliseconds()),
last_rtt_process_time_(clock_->TimeInMilliseconds()),
@@ -101,7 +101,9 @@
rtp_sender_.reset(new RTPSender(
configuration.audio, configuration.clock,
configuration.outgoing_transport, configuration.paced_sender,
- configuration.flexfec_sender,
+ configuration.flexfec_sender
+ ? absl::make_optional(configuration.flexfec_sender->ssrc())
+ : absl::nullopt,
configuration.transport_sequence_number_allocator,
configuration.transport_feedback_callback,
configuration.send_bitrate_observer,
@@ -112,6 +114,14 @@
configuration.populate_network2_timestamp,
configuration.frame_encryptor, configuration.require_frame_encryption,
configuration.extmap_allow_mixed));
+ if (configuration.audio) {
+ audio_ = absl::make_unique<RTPSenderAudio>(clock_, rtp_sender_.get());
+ } else {
+ video_ = absl::make_unique<RTPSenderVideo>(
+ clock_, rtp_sender_.get(), configuration.flexfec_sender,
+ configuration.frame_encryptor,
+ configuration.require_frame_encryption);
+ }
// Make sure rtcp sender use same timestamp offset as rtp sender.
rtcp_sender_.SetTimestampOffset(rtp_sender_->TimestampOffset());
@@ -268,22 +278,21 @@
int frequency,
int channels,
int rate) {
+ RTC_DCHECK(audio_);
rtcp_sender_.SetRtpClockRate(payload_type, frequency);
- RTC_CHECK_EQ(0,
- rtp_sender_->RegisterPayload(payload_name, payload_type,
- frequency, channels, rate));
+ RTC_CHECK_EQ(0, audio_->RegisterAudioPayload(payload_name, payload_type,
+ frequency, channels, rate));
}
void ModuleRtpRtcpImpl::RegisterVideoSendPayload(int payload_type,
const char* payload_name) {
+ RTC_DCHECK(video_);
rtcp_sender_.SetRtpClockRate(payload_type, kVideoPayloadTypeFrequency);
- RTC_CHECK_EQ(0,
- rtp_sender_->RegisterPayload(payload_name, payload_type,
- kVideoPayloadTypeFrequency, 0, 0));
+ video_->RegisterPayloadType(payload_type, payload_name);
}
int32_t ModuleRtpRtcpImpl::DeRegisterSendPayload(const int8_t payload_type) {
- return rtp_sender_->DeRegisterSendPayload(payload_type);
+ return 0;
}
uint32_t ModuleRtpRtcpImpl::StartTimestamp() const {
@@ -446,10 +455,22 @@
expected_retransmission_time_ms = kDefaultExpectedRetransmissionTimeMs;
}
}
- return rtp_sender_->SendOutgoingData(
- frame_type, payload_type, time_stamp, capture_time_ms, payload_data,
- payload_size, fragmentation, rtp_video_header, transport_frame_id_out,
- expected_retransmission_time_ms);
+
+ const uint32_t rtp_timestamp = time_stamp + rtp_sender_->TimestampOffset();
+ if (transport_frame_id_out)
+ *transport_frame_id_out = rtp_timestamp;
+
+ if (audio_) {
+ RTC_DCHECK(fragmentation == nullptr);
+
+ return audio_->SendAudio(frame_type, payload_type, rtp_timestamp,
+ payload_data, payload_size);
+ } else {
+ return video_->SendVideo(frame_type, payload_type, rtp_timestamp,
+ capture_time_ms, payload_data, payload_size,
+ fragmentation, rtp_video_header,
+ expected_retransmission_time_ms);
+ }
}
bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc,
@@ -764,11 +785,11 @@
int32_t ModuleRtpRtcpImpl::SendTelephoneEventOutband(const uint8_t key,
const uint16_t time_ms,
const uint8_t level) {
- return rtp_sender_->SendTelephoneEvent(key, time_ms, level);
+ return audio_ ? audio_->SendTelephoneEvent(key, time_ms, level) : -1;
}
int32_t ModuleRtpRtcpImpl::SetAudioLevel(const uint8_t level_d_bov) {
- return rtp_sender_->SetAudioLevel(level_d_bov);
+ return audio_ ? audio_->SetAudioLevel(level_d_bov) : -1;
}
int32_t ModuleRtpRtcpImpl::SetKeyFrameRequestMethod(
@@ -789,13 +810,18 @@
void ModuleRtpRtcpImpl::SetUlpfecConfig(int red_payload_type,
int ulpfec_payload_type) {
- rtp_sender_->SetUlpfecConfig(red_payload_type, ulpfec_payload_type);
+ RTC_DCHECK(video_);
+ video_->SetUlpfecConfig(red_payload_type, ulpfec_payload_type);
}
bool ModuleRtpRtcpImpl::SetFecParameters(
const FecProtectionParams& delta_params,
const FecProtectionParams& key_params) {
- return rtp_sender_->SetFecParameters(delta_params, key_params);
+ if (!video_) {
+ return false;
+ }
+ video_->SetFecParameters(delta_params, key_params);
+ return true;
}
void ModuleRtpRtcpImpl::SetRemoteSSRC(const uint32_t ssrc) {
@@ -809,13 +835,13 @@
uint32_t* fec_rate,
uint32_t* nack_rate) const {
*total_rate = rtp_sender_->BitrateSent();
- *video_rate = rtp_sender_->VideoBitrateSent();
- *fec_rate = rtp_sender_->FecOverheadRate();
+ *video_rate = video_ ? video_->VideoBitrateSent() : 0;
+ *fec_rate = video_ ? video_->FecOverheadRate() : 0;
*nack_rate = rtp_sender_->NackOverheadRate();
}
uint32_t ModuleRtpRtcpImpl::PacketizationOverheadBps() const {
- return rtp_sender_->PacketizationOverheadBps();
+ return video_ ? video_->PacketizationOverheadBps() : 0;
}
void ModuleRtpRtcpImpl::OnRequestSendReport() {
@@ -843,8 +869,15 @@
void ModuleRtpRtcpImpl::OnReceivedRtcpReportBlocks(
const ReportBlockList& report_blocks) {
- if (rtp_sender_)
- rtp_sender_->OnReceivedRtcpReportBlocks(report_blocks);
+ if (video_) {
+ uint32_t ssrc = SSRC();
+
+ for (const RTCPReportBlock& report_block : report_blocks) {
+ if (ssrc == report_block.source_ssrc) {
+ video_->OnReceivedAck(report_block.extended_highest_sequence_number);
+ }
+ }
+ }
}
bool ModuleRtpRtcpImpl::LastReceivedNTP(
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 61adb31..114e897 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -32,6 +32,8 @@
#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"
@@ -336,13 +338,13 @@
bool TimeToSendFullNackList(int64_t now) const;
std::unique_ptr<RTPSender> rtp_sender_;
+ std::unique_ptr<RTPSenderAudio> audio_;
+ std::unique_ptr<RTPSenderVideo> video_;
RTCPSender rtcp_sender_;
RTCPReceiver rtcp_receiver_;
Clock* const clock_;
- const bool audio_;
-
const RtpKeepAliveConfig keepalive_config_;
int64_t last_bitrate_process_time_;
int64_t last_rtt_process_time_;
diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc
index 7f42a62..ad7e631 100644
--- a/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/modules/rtp_rtcp/source/rtp_sender.cc
@@ -20,14 +20,11 @@
#include "api/array_view.h"
#include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
#include "logging/rtc_event_log/rtc_event_log.h"
-#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "modules/rtp_rtcp/include/rtp_cvo.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
-#include "modules/rtp_rtcp/source/rtp_sender_audio.h"
-#include "modules/rtp_rtcp/source/rtp_sender_video.h"
#include "modules/rtp_rtcp/source/time_util.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/checks.h"
@@ -35,7 +32,6 @@
#include "rtc_base/numerics/safe_minmax.h"
#include "rtc_base/rate_limiter.h"
#include "rtc_base/time_utils.h"
-#include "rtc_base/trace_event.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
@@ -87,21 +83,6 @@
RtpGenericFrameDescriptorExtension::kMaxSizeBytes},
};
-const char* FrameTypeToString(FrameType frame_type) {
- switch (frame_type) {
- case kEmptyFrame:
- return "empty";
- case kAudioFrameSpeech:
- return "audio_speech";
- case kAudioFrameCN:
- return "audio_cn";
- case kVideoFrameKey:
- return "video_key";
- case kVideoFrameDelta:
- return "video_delta";
- }
- return "";
-}
} // namespace
RTPSender::RTPSender(
@@ -109,7 +90,7 @@
Clock* clock,
Transport* transport,
RtpPacketSender* paced_sender,
- FlexfecSender* flexfec_sender,
+ absl::optional<uint32_t> flexfec_ssrc,
TransportSequenceNumberAllocator* sequence_number_allocator,
TransportFeedbackObserver* transport_feedback_observer,
BitrateStatisticsObserver* bitrate_callback,
@@ -127,13 +108,7 @@
clock_delta_ms_(clock_->TimeInMilliseconds() - rtc::TimeMillis()),
random_(clock_->TimeInMicroseconds()),
audio_configured_(audio),
- audio_(audio ? new RTPSenderAudio(clock, this) : nullptr),
- video_(audio ? nullptr
- : new RTPSenderVideo(clock,
- this,
- flexfec_sender,
- frame_encryptor,
- require_frame_encryption)),
+ flexfec_ssrc_(flexfec_ssrc),
paced_sender_(paced_sender),
transport_sequence_number_allocator_(sequence_number_allocator),
transport_feedback_observer_(transport_feedback_observer),
@@ -180,7 +155,7 @@
// Store FlexFEC packets in the packet history data structure, so they can
// be found when paced.
- if (flexfec_sender) {
+ if (flexfec_ssrc_) {
flexfec_packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStore,
kMinFlexfecPacketsToStoreForPacing);
@@ -216,29 +191,11 @@
1000);
}
-uint32_t RTPSender::VideoBitrateSent() const {
- if (video_) {
- return video_->VideoBitrateSent();
- }
- return 0;
-}
-
-uint32_t RTPSender::FecOverheadRate() const {
- if (video_) {
- return video_->FecOverheadRate();
- }
- return 0;
-}
-
uint32_t RTPSender::NackOverheadRate() const {
rtc::CritScope cs(&statistics_crit_);
return nack_bitrate_sent_.Rate(clock_->TimeInMilliseconds()).value_or(0);
}
-uint32_t RTPSender::PacketizationOverheadBps() const {
- return video_ ? video_->PacketizationOverheadBps() : 0;
-}
-
void RTPSender::SetExtmapAllowMixed(bool extmap_allow_mixed) {
rtc::CritScope lock(&send_critsect_);
rtp_header_extension_map_.SetExtmapAllowMixed(extmap_allow_mixed);
@@ -265,29 +222,6 @@
return rtp_header_extension_map_.Deregister(type);
}
-int32_t RTPSender::RegisterPayload(absl::string_view payload_name,
- int8_t payload_number,
- uint32_t frequency,
- size_t channels,
- uint32_t rate) {
- rtc::CritScope lock(&send_critsect_);
-
- int32_t ret_val = 0;
- if (audio_configured_) {
- // TODO(mflodman): Change to CreateAudioPayload and make static.
- ret_val = audio_->RegisterAudioPayload(payload_name, payload_number,
- frequency, channels, rate);
- } else {
- video_->RegisterPayloadType(payload_number, payload_name);
- }
-
- return ret_val;
-}
-
-int32_t RTPSender::DeRegisterSendPayload(int8_t /* payload_type */) {
- return 0;
-}
-
void RTPSender::SetMaxRtpPacketSize(size_t max_packet_size) {
RTC_DCHECK_GE(max_packet_size, 100);
RTC_DCHECK_LE(max_packet_size, IP_PACKET_SIZE);
@@ -333,67 +267,6 @@
rtx_payload_type_map_[associated_payload_type] = payload_type;
}
-bool RTPSender::SendOutgoingData(FrameType frame_type,
- int8_t payload_type,
- uint32_t capture_timestamp,
- int64_t capture_time_ms,
- const uint8_t* payload_data,
- size_t payload_size,
- const RTPFragmentationHeader* fragmentation,
- const RTPVideoHeader* rtp_header,
- uint32_t* transport_frame_id_out,
- int64_t expected_retransmission_time_ms) {
- uint16_t sequence_number;
- uint32_t rtp_timestamp;
- {
- // Drop this packet if we're not sending media packets.
- rtc::CritScope lock(&send_critsect_);
- RTC_DCHECK(ssrc_);
-
- sequence_number = sequence_number_;
- rtp_timestamp = timestamp_offset_ + capture_timestamp;
- if (transport_frame_id_out)
- *transport_frame_id_out = rtp_timestamp;
- if (!sending_media_)
- return true;
- }
- switch (frame_type) {
- case kAudioFrameSpeech:
- case kAudioFrameCN:
- RTC_CHECK(audio_configured_);
- break;
- case kVideoFrameKey:
- case kVideoFrameDelta:
- RTC_CHECK(!audio_configured_);
- break;
- case kEmptyFrame:
- break;
- }
-
- bool result;
- if (audio_configured_) {
- TRACE_EVENT_ASYNC_STEP1("webrtc", "Audio", rtp_timestamp, "Send", "type",
- FrameTypeToString(frame_type));
- // The only known way to produce of RTPFragmentationHeader for audio is
- // to use the AudioCodingModule directly.
- RTC_DCHECK(fragmentation == nullptr);
- result = audio_->SendAudio(frame_type, payload_type, rtp_timestamp,
- payload_data, payload_size);
- } else {
- TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", capture_time_ms, "Send", "type",
- FrameTypeToString(frame_type));
- if (frame_type == kEmptyFrame)
- return true;
-
- result = video_->SendVideo(frame_type, payload_type, rtp_timestamp,
- capture_time_ms, payload_data, payload_size,
- fragmentation, rtp_header,
- expected_retransmission_time_ms);
- }
-
- return result;
-}
-
size_t RTPSender::TrySendRedundantPayloads(size_t bytes_to_send,
const PacedPacketInfo& pacing_info) {
{
@@ -643,26 +516,6 @@
}
}
-void RTPSender::OnReceivedRtcpReportBlocks(
- const ReportBlockList& report_blocks) {
- if (!video_) {
- return;
- }
- uint32_t ssrc;
- {
- rtc::CritScope lock(&send_critsect_);
- if (!ssrc_)
- return;
- ssrc = *ssrc_;
- }
-
- for (const RTCPReportBlock& report_block : report_blocks) {
- if (ssrc == report_block.source_ssrc) {
- video_->OnReceivedAck(report_block.extended_highest_sequence_number);
- }
- }
-}
-
// Called from pacer when we can send the packet.
bool RTPSender::TimeToSendPacket(uint32_t ssrc,
uint16_t sequence_number,
@@ -809,29 +662,14 @@
RTC_DCHECK(packet);
int64_t now_ms = clock_->TimeInMilliseconds();
- if (video_) {
- BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoTotBitrate_kbps", now_ms,
- ActualSendBitrateKbit(), packet->Ssrc());
- BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoFecBitrate_kbps", now_ms,
- FecOverheadRate() / 1000, packet->Ssrc());
- BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoNackBitrate_kbps", now_ms,
- NackOverheadRate() / 1000, packet->Ssrc());
- } else {
- BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioTotBitrate_kbps", now_ms,
- ActualSendBitrateKbit(), packet->Ssrc());
- BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioNackBitrate_kbps", now_ms,
- NackOverheadRate() / 1000, packet->Ssrc());
- }
-
uint32_t ssrc = packet->Ssrc();
- absl::optional<uint32_t> flexfec_ssrc = FlexfecSsrc();
if (paced_sender_) {
uint16_t seq_no = packet->SequenceNumber();
// Correct offset between implementations of millisecond time stamps in
// TickTime and Clock.
int64_t corrected_time_ms = packet->capture_time_ms() + clock_delta_ms_;
size_t payload_length = packet->payload_size();
- if (ssrc == flexfec_ssrc) {
+ if (ssrc == FlexfecSsrc()) {
// Store FlexFEC packets in the history here, so they can be found
// when the pacer calls TimeToSendPacket.
flexfec_packet_history_.PutRtpPacket(std::move(packet), storage,
@@ -1164,10 +1002,7 @@
}
absl::optional<uint32_t> RTPSender::FlexfecSsrc() const {
- if (video_) {
- return video_->FlexfecSsrc();
- }
- return absl::nullopt;
+ return flexfec_ssrc_;
}
void RTPSender::SetCsrcs(const std::vector<uint32_t>& csrcs) {
@@ -1187,34 +1022,6 @@
return sequence_number_;
}
-// Audio.
-int32_t RTPSender::SendTelephoneEvent(uint8_t key,
- uint16_t time_ms,
- uint8_t level) {
- if (!audio_configured_) {
- return -1;
- }
- return audio_->SendTelephoneEvent(key, time_ms, level);
-}
-
-int32_t RTPSender::SetAudioLevel(uint8_t level_d_bov) {
- return audio_->SetAudioLevel(level_d_bov);
-}
-
-void RTPSender::SetUlpfecConfig(int red_payload_type, int ulpfec_payload_type) {
- RTC_DCHECK(!audio_configured_);
- video_->SetUlpfecConfig(red_payload_type, ulpfec_payload_type);
-}
-
-bool RTPSender::SetFecParameters(const FecProtectionParams& delta_params,
- const FecProtectionParams& key_params) {
- if (audio_configured_) {
- return false;
- }
- video_->SetFecParameters(delta_params, key_params);
- return true;
-}
-
static std::unique_ptr<RtpPacketToSend> CreateRtxPacket(
const RtpPacketToSend& packet,
RtpHeaderExtensionMap* extension_map) {
diff --git a/modules/rtp_rtcp/source/rtp_sender.h b/modules/rtp_rtcp/source/rtp_sender.h
index 401a8ff..622f41a 100644
--- a/modules/rtp_rtcp/source/rtp_sender.h
+++ b/modules/rtp_rtcp/source/rtp_sender.h
@@ -41,8 +41,6 @@
class RateLimiter;
class RtcEventLog;
class RtpPacketToSend;
-class RTPSenderAudio;
-class RTPSenderVideo;
class RTPSender {
public:
@@ -50,9 +48,7 @@
Clock* clock,
Transport* transport,
RtpPacketSender* paced_sender,
- // TODO(brandtr): Remove |flexfec_sender| when that is hooked up
- // to PacedSender instead.
- FlexfecSender* flexfec_sender,
+ absl::optional<uint32_t> flexfec_ssrc,
TransportSequenceNumberAllocator* sequence_number_allocator,
TransportFeedbackObserver* transport_feedback_callback,
BitrateStatisticsObserver* bitrate_callback,
@@ -72,18 +68,7 @@
uint16_t ActualSendBitrateKbit() const;
- uint32_t VideoBitrateSent() const;
- uint32_t FecOverheadRate() const;
uint32_t NackOverheadRate() const;
- uint32_t PacketizationOverheadBps() const;
-
- int32_t RegisterPayload(absl::string_view payload_name,
- const int8_t payload_type,
- const uint32_t frequency,
- const size_t channels,
- const uint32_t rate);
-
- int32_t DeRegisterSendPayload(const int8_t payload_type);
void SetSendingMediaStatus(bool enabled);
bool SendingMedia() const;
@@ -109,17 +94,6 @@
void SetMaxRtpPacketSize(size_t max_packet_size);
- 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_header,
- uint32_t* transport_frame_id_out,
- int64_t expected_retransmission_time_ms);
-
void SetExtmapAllowMixed(bool extmap_allow_mixed);
// RTP header extension
@@ -145,10 +119,6 @@
int32_t ReSendPacket(uint16_t packet_id);
- // Feedback to decide when to stop sending the playout delay and MID header
- // extensions.
- void OnReceivedRtcpReportBlocks(const ReportBlockList& report_blocks);
-
// RTX.
void SetRtxStatus(int mode);
int RtxStatus() const;
@@ -187,21 +157,6 @@
StorageType storage,
RtpPacketSender::Priority priority);
- // Audio.
-
- // Send a DTMF tone using RFC 2833 (4733).
- int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
-
- // Store the audio level in d_bov for
- // header-extension-for-audio-level-indication.
- int32_t SetAudioLevel(uint8_t level_d_bov);
-
- // ULPFEC.
- void SetUlpfecConfig(int red_payload_type, int ulpfec_payload_type);
-
- bool SetFecParameters(const FecProtectionParams& delta_params,
- const FecProtectionParams& key_params);
-
// Called on update of RTP statistics.
void RegisterRtpStatisticsCallback(StreamDataCountersCallback* callback);
StreamDataCountersCallback* GetRtpStatisticsCallback() const;
@@ -269,8 +224,8 @@
Random random_ RTC_GUARDED_BY(send_critsect_);
const bool audio_configured_;
- const std::unique_ptr<RTPSenderAudio> audio_;
- const std::unique_ptr<RTPSenderVideo> video_;
+
+ const absl::optional<uint32_t> flexfec_ssrc_;
RtpPacketSender* const paced_sender_;
TransportSequenceNumberAllocator* const transport_sequence_number_allocator_;
diff --git a/modules/rtp_rtcp/source/rtp_sender_audio.cc b/modules/rtp_rtcp/source/rtp_sender_audio.cc
index 9acc098..56d0884 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -16,6 +16,7 @@
#include "absl/strings/match.h"
#include "api/audio_codecs/audio_format.h"
+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
@@ -27,6 +28,24 @@
namespace webrtc {
+namespace {
+
+const char* FrameTypeToString(FrameType frame_type) {
+ switch (frame_type) {
+ case kEmptyFrame:
+ return "empty";
+ case kAudioFrameSpeech:
+ return "audio_speech";
+ case kAudioFrameCN:
+ return "audio_cn";
+ default:
+ RTC_NOTREACHED();
+ return "";
+ }
+}
+
+} // namespace
+
RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtp_sender)
: clock_(clock), rtp_sender_(rtp_sender) {}
@@ -115,6 +134,12 @@
uint32_t rtp_timestamp,
const uint8_t* payload_data,
size_t payload_size) {
+ RTC_DCHECK(frame_type == kAudioFrameSpeech || frame_type == kAudioFrameCN ||
+ frame_type == kEmptyFrame);
+
+ TRACE_EVENT_ASYNC_STEP1("webrtc", "Audio", rtp_timestamp, "Send", "type",
+ FrameTypeToString(frame_type));
+
// From RFC 4733:
// A source has wide latitude as to how often it sends event updates. A
// natural interval is the spacing between non-event audio packets. [...]
@@ -233,7 +258,7 @@
TRACE_EVENT_ASYNC_END2("webrtc", "Audio", rtp_timestamp, "timestamp",
packet->Timestamp(), "seqnum",
packet->SequenceNumber());
- bool send_result = rtp_sender_->SendToNetwork(
+ bool send_result = LogAndSendToNetwork(
std::move(packet), kAllowRetransmission, RtpPacketSender::kHighPriority);
if (first_packet_sent_()) {
RTC_LOG(LS_INFO) << "First audio RTP packet sent to pacer";
@@ -317,11 +342,29 @@
dtmfbuffer[1] = E | R | volume;
ByteWriter<uint16_t>::WriteBigEndian(dtmfbuffer + 2, duration);
- result = rtp_sender_->SendToNetwork(std::move(packet), kAllowRetransmission,
- RtpPacketSender::kHighPriority);
+ result = LogAndSendToNetwork(std::move(packet), kAllowRetransmission,
+ RtpPacketSender::kHighPriority);
send_count--;
} while (send_count > 0 && result);
return result;
}
+
+bool RTPSenderAudio::LogAndSendToNetwork(
+ std::unique_ptr<RtpPacketToSend> packet,
+ StorageType storage,
+ RtpPacketSender::Priority priority) {
+#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
+ int64_t now_ms = clock_->TimeInMilliseconds();
+ BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioTotBitrate_kbps", now_ms,
+ rtp_sender_->ActualSendBitrateKbit(),
+ packet->Ssrc());
+ BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioNackBitrate_kbps", now_ms,
+ rtp_sender_->NackOverheadRate() / 1000,
+ packet->Ssrc());
+#endif
+
+ return rtp_sender_->SendToNetwork(std::move(packet), storage, priority);
+}
+
} // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_sender_audio.h b/modules/rtp_rtcp/source/rtp_sender_audio.h
index f002023..fa58943 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio.h
+++ b/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -14,6 +14,8 @@
#include <stddef.h>
#include <stdint.h>
+#include <memory>
+
#include "absl/strings/string_view.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/rtp_rtcp/source/dtmf_queue.h"
@@ -61,6 +63,10 @@
bool MarkerBit(FrameType frame_type, int8_t payload_type);
private:
+ bool LogAndSendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
+ StorageType storage,
+ RtpPacketSender::Priority priority);
+
Clock* const clock_ = nullptr;
RTPSender* const rtp_sender_ = nullptr;
diff --git a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
index b1c46c1..7f9d72f 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
@@ -64,7 +64,7 @@
&fake_clock_,
&transport_,
nullptr,
- nullptr,
+ absl::nullopt,
nullptr,
nullptr,
nullptr,
diff --git a/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index b13875a..714fac7 100644
--- a/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -184,7 +184,7 @@
void SetUpRtpSender(bool pacer, bool populate_network2) {
rtp_sender_.reset(new RTPSender(
false, &fake_clock_, &transport_, pacer ? &mock_paced_sender_ : nullptr,
- nullptr, &seq_num_allocator_, nullptr, nullptr, nullptr,
+ absl::nullopt, &seq_num_allocator_, nullptr, nullptr, nullptr,
&mock_rtc_event_log_, &send_packet_observer_,
&retransmission_rate_limiter_, nullptr, populate_network2, nullptr,
false, false));
@@ -324,9 +324,10 @@
MockTransport transport;
const bool kEnableAudio = true;
rtp_sender_.reset(new RTPSender(
- kEnableAudio, &fake_clock_, &transport, &mock_paced_sender_, nullptr,
- nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_, nullptr,
- &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
+ kEnableAudio, &fake_clock_, &transport, &mock_paced_sender_,
+ absl::nullopt, nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
+ nullptr, &retransmission_rate_limiter_, nullptr, false, nullptr, false,
+ false));
rtp_sender_->SetTimestampOffset(0);
rtp_sender_->SetSSRC(kSsrc);
@@ -370,10 +371,10 @@
constexpr int kRtpOverheadBytesPerPacket = 12 + 8;
testing::NiceMock<MockOverheadObserver> mock_overhead_observer;
rtp_sender_.reset(new RTPSender(
- false, &fake_clock_, &transport_, nullptr, nullptr, &seq_num_allocator_,
- &feedback_observer_, nullptr, nullptr, &mock_rtc_event_log_, nullptr,
- &retransmission_rate_limiter_, &mock_overhead_observer, false, nullptr,
- false, false));
+ false, &fake_clock_, &transport_, nullptr, absl::nullopt,
+ &seq_num_allocator_, &feedback_observer_, nullptr, nullptr,
+ &mock_rtc_event_log_, nullptr, &retransmission_rate_limiter_,
+ &mock_overhead_observer, false, nullptr, false, false));
rtp_sender_->SetSSRC(kSsrc);
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
@@ -397,10 +398,10 @@
TEST_P(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) {
rtp_sender_.reset(new RTPSender(
- false, &fake_clock_, &transport_, nullptr, nullptr, &seq_num_allocator_,
- &feedback_observer_, nullptr, nullptr, &mock_rtc_event_log_,
- &send_packet_observer_, &retransmission_rate_limiter_, nullptr, false,
- nullptr, false, false));
+ false, &fake_clock_, &transport_, nullptr, absl::nullopt,
+ &seq_num_allocator_, &feedback_observer_, nullptr, nullptr,
+ &mock_rtc_event_log_, &send_packet_observer_,
+ &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
rtp_sender_->SetSSRC(kSsrc);
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
@@ -429,10 +430,10 @@
TEST_P(RtpSenderTestWithoutPacer, PacketOptionsNoRetransmission) {
rtp_sender_.reset(new RTPSender(
- false, &fake_clock_, &transport_, nullptr, nullptr, &seq_num_allocator_,
- &feedback_observer_, nullptr, nullptr, &mock_rtc_event_log_,
- &send_packet_observer_, &retransmission_rate_limiter_, nullptr, false,
- nullptr, false, false));
+ false, &fake_clock_, &transport_, nullptr, absl::nullopt,
+ &seq_num_allocator_, &feedback_observer_, nullptr, nullptr,
+ &mock_rtc_event_log_, &send_packet_observer_,
+ &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
rtp_sender_->SetSSRC(kSsrc);
SendGenericPacket();
@@ -484,18 +485,20 @@
TEST_P(RtpSenderTestWithoutPacer, OnSendSideDelayUpdated) {
testing::StrictMock<MockSendSideDelayObserver> send_side_delay_observer_;
rtp_sender_.reset(new RTPSender(
- false, &fake_clock_, &transport_, nullptr, nullptr, nullptr, nullptr,
- nullptr, &send_side_delay_observer_, &mock_rtc_event_log_, nullptr,
- nullptr, nullptr, false, nullptr, false, false));
+ false, &fake_clock_, &transport_, nullptr, absl::nullopt, nullptr,
+ nullptr, nullptr, &send_side_delay_observer_, &mock_rtc_event_log_,
+ nullptr, nullptr, nullptr, false, nullptr, false, false));
rtp_sender_->SetSSRC(kSsrc);
+ RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(), nullptr,
+ nullptr, false);
const uint8_t kPayloadType = 127;
- const uint32_t kCaptureTimeMsToRtpTimestamp = 90; // 90 kHz clock
const char payload_name[] = "GENERIC";
+
+ rtp_sender_video.RegisterPayloadType(kPayloadType, payload_name);
+
+ const uint32_t kCaptureTimeMsToRtpTimestamp = 90; // 90 kHz clock
RTPVideoHeader video_header;
- EXPECT_EQ(0, rtp_sender_->RegisterPayload(payload_name, kPayloadType,
- 1000 * kCaptureTimeMsToRtpTimestamp,
- 0, 1500));
// Send packet with 10 ms send-side delay. The average and max should be 10
// ms.
@@ -503,10 +506,10 @@
.Times(1);
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
fake_clock_.AdvanceTimeMilliseconds(10);
- EXPECT_TRUE(rtp_sender_->SendOutgoingData(
+ EXPECT_TRUE(rtp_sender_video.SendVideo(
kVideoFrameKey, kPayloadType,
capture_time_ms * kCaptureTimeMsToRtpTimestamp, capture_time_ms,
- kPayloadData, sizeof(kPayloadData), nullptr, &video_header, nullptr,
+ kPayloadData, sizeof(kPayloadData), nullptr, &video_header,
kDefaultExpectedRetransmissionTimeMs));
// Send another packet with 20 ms delay. The average
@@ -514,10 +517,10 @@
EXPECT_CALL(send_side_delay_observer_, SendSideDelayUpdated(15, 20, kSsrc))
.Times(1);
fake_clock_.AdvanceTimeMilliseconds(10);
- EXPECT_TRUE(rtp_sender_->SendOutgoingData(
+ EXPECT_TRUE(rtp_sender_video.SendVideo(
kVideoFrameKey, kPayloadType,
capture_time_ms * kCaptureTimeMsToRtpTimestamp, capture_time_ms,
- kPayloadData, sizeof(kPayloadData), nullptr, &video_header, nullptr,
+ kPayloadData, sizeof(kPayloadData), nullptr, &video_header,
kDefaultExpectedRetransmissionTimeMs));
// Send another packet at the same time, which replaces the last packet.
@@ -526,10 +529,10 @@
EXPECT_CALL(send_side_delay_observer_, SendSideDelayUpdated(5, 10, kSsrc))
.Times(1);
capture_time_ms = fake_clock_.TimeInMilliseconds();
- EXPECT_TRUE(rtp_sender_->SendOutgoingData(
+ EXPECT_TRUE(rtp_sender_video.SendVideo(
kVideoFrameKey, kPayloadType,
capture_time_ms * kCaptureTimeMsToRtpTimestamp, capture_time_ms,
- kPayloadData, sizeof(kPayloadData), nullptr, &video_header, nullptr,
+ kPayloadData, sizeof(kPayloadData), nullptr, &video_header,
kDefaultExpectedRetransmissionTimeMs));
// Send a packet 1 second later. The earlier packets should have timed
@@ -539,10 +542,10 @@
fake_clock_.AdvanceTimeMilliseconds(1);
EXPECT_CALL(send_side_delay_observer_, SendSideDelayUpdated(1, 1, kSsrc))
.Times(1);
- EXPECT_TRUE(rtp_sender_->SendOutgoingData(
+ EXPECT_TRUE(rtp_sender_video.SendVideo(
kVideoFrameKey, kPayloadType,
capture_time_ms * kCaptureTimeMsToRtpTimestamp, capture_time_ms,
- kPayloadData, sizeof(kPayloadData), nullptr, &video_header, nullptr,
+ kPayloadData, sizeof(kPayloadData), nullptr, &video_header,
kDefaultExpectedRetransmissionTimeMs));
}
@@ -561,7 +564,7 @@
TEST_P(RtpSenderTest, SendsPacketsWithTransportSequenceNumber) {
rtp_sender_.reset(new RTPSender(
- false, &fake_clock_, &transport_, &mock_paced_sender_, nullptr,
+ false, &fake_clock_, &transport_, &mock_paced_sender_, absl::nullopt,
&seq_num_allocator_, &feedback_observer_, nullptr, nullptr,
&mock_rtc_event_log_, &send_packet_observer_,
&retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
@@ -946,7 +949,7 @@
TEST_P(RtpSenderTest, OnSendPacketNotUpdatedWithoutSeqNumAllocator) {
rtp_sender_.reset(new RTPSender(
- false, &fake_clock_, &transport_, &mock_paced_sender_, nullptr,
+ false, &fake_clock_, &transport_, &mock_paced_sender_, absl::nullopt,
nullptr /* TransportSequenceNumberAllocator */, nullptr, nullptr, nullptr,
nullptr, &send_packet_observer_, &retransmission_rate_limiter_, nullptr,
false, nullptr, false, false));
@@ -973,8 +976,8 @@
TEST_P(RtpSenderTest, SendRedundantPayloads) {
MockTransport transport;
rtp_sender_.reset(new RTPSender(
- false, &fake_clock_, &transport, &mock_paced_sender_, nullptr, nullptr,
- nullptr, nullptr, nullptr, &mock_rtc_event_log_, nullptr,
+ false, &fake_clock_, &transport, &mock_paced_sender_, absl::nullopt,
+ nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_, nullptr,
&retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetSSRC(kSsrc);
@@ -1050,15 +1053,16 @@
TEST_P(RtpSenderTestWithoutPacer, SendGenericVideo) {
const char payload_name[] = "GENERIC";
const uint8_t payload_type = 127;
- ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
- 0, 1500));
+ RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(), nullptr,
+ nullptr, false);
+ rtp_sender_video.RegisterPayloadType(payload_type, payload_name);
uint8_t payload[] = {47, 11, 32, 93, 89};
// Send keyframe
RTPVideoHeader video_header;
- ASSERT_TRUE(rtp_sender_->SendOutgoingData(
+ ASSERT_TRUE(rtp_sender_video.SendVideo(
kVideoFrameKey, payload_type, 1234, 4321, payload, sizeof(payload),
- nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
+ nullptr, &video_header, kDefaultExpectedRetransmissionTimeMs));
auto sent_payload = transport_.last_sent_packet().payload();
uint8_t generic_header = sent_payload[0];
@@ -1071,9 +1075,9 @@
payload[1] = 42;
payload[4] = 13;
- ASSERT_TRUE(rtp_sender_->SendOutgoingData(
+ ASSERT_TRUE(rtp_sender_video.SendVideo(
kVideoFrameDelta, payload_type, 1234, 4321, payload, sizeof(payload),
- nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
+ nullptr, &video_header, kDefaultExpectedRetransmissionTimeMs));
sent_payload = transport_.last_sent_packet().payload();
generic_header = sent_payload[0];
@@ -1096,7 +1100,7 @@
// Reset |rtp_sender_| to use FlexFEC.
rtp_sender_.reset(new RTPSender(
- false, &fake_clock_, &transport_, &mock_paced_sender_, &flexfec_sender,
+ false, &fake_clock_, &transport_, &mock_paced_sender_, kFlexfecSsrc,
&seq_num_allocator_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
&send_packet_observer_, &retransmission_rate_limiter_, nullptr, false,
nullptr, false, false));
@@ -1167,10 +1171,10 @@
// Reset |rtp_sender_| to use FlexFEC.
rtp_sender_.reset(new RTPSender(
- false, &fake_clock_, &transport_, &mock_paced_sender_, &flexfec_sender,
- &seq_num_allocator_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
- &send_packet_observer_, &retransmission_rate_limiter_, nullptr, false,
- nullptr, false, false));
+ false, &fake_clock_, &transport_, &mock_paced_sender_,
+ flexfec_sender.ssrc(), &seq_num_allocator_, nullptr, nullptr, nullptr,
+ &mock_rtc_event_log_, &send_packet_observer_,
+ &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
rtp_sender_->SetSSRC(kMediaSsrc);
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetStorePacketsStatus(true, 10);
@@ -1264,7 +1268,7 @@
// Reset |rtp_sender_| to use FlexFEC.
rtp_sender_.reset(new RTPSender(
- false, &fake_clock_, &transport_, nullptr, &flexfec_sender,
+ false, &fake_clock_, &transport_, nullptr, flexfec_sender.ssrc(),
&seq_num_allocator_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
&send_packet_observer_, &retransmission_rate_limiter_, nullptr, false,
nullptr, false, false));
@@ -1391,10 +1395,10 @@
// Reset |rtp_sender_| to use FlexFEC.
rtp_sender_.reset(new RTPSender(
- false, &fake_clock_, &transport_, &mock_paced_sender_, &flexfec_sender,
- &seq_num_allocator_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
- &send_packet_observer_, &retransmission_rate_limiter_, nullptr, false,
- nullptr, false, false));
+ false, &fake_clock_, &transport_, &mock_paced_sender_,
+ flexfec_sender.ssrc(), &seq_num_allocator_, nullptr, nullptr, nullptr,
+ &mock_rtc_event_log_, &send_packet_observer_,
+ &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
rtp_sender_->SetSSRC(kMediaSsrc);
rtp_sender_->SetSequenceNumber(kSeqNum);
@@ -1460,11 +1464,17 @@
uint32_t retransmit_bitrate_;
} callback;
rtp_sender_.reset(new RTPSender(
- false, &fake_clock_, &transport_, nullptr, nullptr, nullptr, nullptr,
- &callback, nullptr, nullptr, nullptr, &retransmission_rate_limiter_,
- nullptr, false, nullptr, false, false));
+ false, &fake_clock_, &transport_, nullptr, absl::nullopt, nullptr,
+ nullptr, &callback, nullptr, nullptr, nullptr,
+ &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
rtp_sender_->SetSSRC(kSsrc);
+ RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(), nullptr,
+ nullptr, false);
+ const char payload_name[] = "GENERIC";
+ const uint8_t payload_type = 127;
+ rtp_sender_video.RegisterPayloadType(payload_type, payload_name);
+
// Simulate kNumPackets sent with kPacketInterval ms intervals, with the
// number of packets selected so that we fill (but don't overflow) the one
// second averaging window.
@@ -1475,10 +1485,6 @@
// Overhead = 12 bytes RTP header + 1 byte generic header.
const uint32_t kPacketOverhead = 13;
- const char payload_name[] = "GENERIC";
- const uint8_t payload_type = 127;
- ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
- 0, 1500));
uint8_t payload[] = {47, 11, 32, 93, 89};
rtp_sender_->SetStorePacketsStatus(true, 1);
uint32_t ssrc = rtp_sender_->SSRC();
@@ -1489,9 +1495,9 @@
// Send a few frames.
RTPVideoHeader video_header;
for (uint32_t i = 0; i < kNumPackets; ++i) {
- ASSERT_TRUE(rtp_sender_->SendOutgoingData(
+ ASSERT_TRUE(rtp_sender_video.SendVideo(
kVideoFrameKey, payload_type, 1234, 4321, payload, sizeof(payload),
- nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
+ nullptr, &video_header, kDefaultExpectedRetransmissionTimeMs));
fake_clock_.AdvanceTimeMilliseconds(kPacketInterval);
}
@@ -1548,8 +1554,9 @@
const uint8_t kUlpfecPayloadType = 97;
const char payload_name[] = "GENERIC";
const uint8_t payload_type = 127;
- ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
- 0, 1500));
+ RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(), nullptr,
+ nullptr, false);
+ rtp_sender_video.RegisterPayloadType(payload_type, payload_name);
uint8_t payload[] = {47, 11, 32, 93, 89};
rtp_sender_->SetStorePacketsStatus(true, 1);
uint32_t ssrc = rtp_sender_->SSRC();
@@ -1558,9 +1565,9 @@
// Send a frame.
RTPVideoHeader video_header;
- ASSERT_TRUE(rtp_sender_->SendOutgoingData(
+ ASSERT_TRUE(rtp_sender_video.SendVideo(
kVideoFrameKey, payload_type, 1234, 4321, payload, sizeof(payload),
- nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
+ nullptr, &video_header, kDefaultExpectedRetransmissionTimeMs));
StreamDataCounters expected;
expected.transmitted.payload_bytes = 6;
expected.transmitted.header_bytes = 12;
@@ -1594,15 +1601,15 @@
callback.Matches(ssrc, expected);
// Send ULPFEC.
- rtp_sender_->SetUlpfecConfig(kRedPayloadType, kUlpfecPayloadType);
+ rtp_sender_video.SetUlpfecConfig(kRedPayloadType, kUlpfecPayloadType);
FecProtectionParams fec_params;
fec_params.fec_mask_type = kFecMaskRandom;
fec_params.fec_rate = 1;
fec_params.max_fec_frames = 1;
- rtp_sender_->SetFecParameters(fec_params, fec_params);
- ASSERT_TRUE(rtp_sender_->SendOutgoingData(
+ rtp_sender_video.SetFecParameters(fec_params, fec_params);
+ ASSERT_TRUE(rtp_sender_video.SendVideo(
kVideoFrameDelta, payload_type, 1234, 4321, payload, sizeof(payload),
- nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
+ nullptr, &video_header, kDefaultExpectedRetransmissionTimeMs));
expected.transmitted.payload_bytes = 40;
expected.transmitted.header_bytes = 60;
expected.transmitted.packets = 5;
@@ -1613,22 +1620,14 @@
}
TEST_P(RtpSenderTestWithoutPacer, BytesReportedCorrectly) {
- const char* kPayloadName = "GENERIC";
+ // XXX const char* kPayloadName = "GENERIC";
const uint8_t kPayloadType = 127;
rtp_sender_->SetSSRC(1234);
rtp_sender_->SetRtxSsrc(4321);
rtp_sender_->SetRtxPayloadType(kPayloadType - 1, kPayloadType);
rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
- ASSERT_EQ(0, rtp_sender_->RegisterPayload(kPayloadName, kPayloadType, 90000,
- 0, 1500));
- uint8_t payload[] = {47, 11, 32, 93, 89};
-
- RTPVideoHeader video_header;
- ASSERT_TRUE(rtp_sender_->SendOutgoingData(
- kVideoFrameKey, kPayloadType, 1234, 4321, payload, sizeof(payload),
- nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
-
+ SendGenericPacket();
// Will send 2 full-size padding packets.
rtp_sender_->TimeToSendPadding(1, PacedPacketInfo());
rtp_sender_->TimeToSendPadding(1, PacedPacketInfo());
@@ -1637,9 +1636,9 @@
StreamDataCounters rtx_stats;
rtp_sender_->GetDataCounters(&rtp_stats, &rtx_stats);
- // Payload + 1-byte generic header.
+ // Payload
EXPECT_GT(rtp_stats.first_packet_time_ms, -1);
- EXPECT_EQ(rtp_stats.transmitted.payload_bytes, sizeof(payload) + 1);
+ EXPECT_EQ(rtp_stats.transmitted.payload_bytes, sizeof(kPayloadData));
EXPECT_EQ(rtp_stats.transmitted.header_bytes, 12u);
EXPECT_EQ(rtp_stats.transmitted.padding_bytes, 0u);
EXPECT_EQ(rtx_stats.transmitted.payload_bytes, 0u);
@@ -1694,10 +1693,11 @@
TEST_P(RtpSenderTest, OnOverheadChanged) {
MockOverheadObserver mock_overhead_observer;
- rtp_sender_.reset(new RTPSender(
- false, &fake_clock_, &transport_, nullptr, nullptr, nullptr, nullptr,
- nullptr, nullptr, nullptr, nullptr, &retransmission_rate_limiter_,
- &mock_overhead_observer, false, nullptr, false, false));
+ rtp_sender_.reset(
+ new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt,
+ nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
+ &retransmission_rate_limiter_, &mock_overhead_observer,
+ false, nullptr, false, false));
rtp_sender_->SetSSRC(kSsrc);
// RTP overhead is 12B.
@@ -1715,10 +1715,11 @@
TEST_P(RtpSenderTest, DoesNotUpdateOverheadOnEqualSize) {
MockOverheadObserver mock_overhead_observer;
- rtp_sender_.reset(new RTPSender(
- false, &fake_clock_, &transport_, nullptr, nullptr, nullptr, nullptr,
- nullptr, nullptr, nullptr, nullptr, &retransmission_rate_limiter_,
- &mock_overhead_observer, false, nullptr, false, false));
+ rtp_sender_.reset(
+ new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt,
+ nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
+ &retransmission_rate_limiter_, &mock_overhead_observer,
+ false, nullptr, false, false));
rtp_sender_->SetSSRC(kSsrc);
EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(_)).Times(1);
@@ -1729,7 +1730,7 @@
TEST_P(RtpSenderTest, SendsKeepAlive) {
MockTransport transport;
rtp_sender_.reset(new RTPSender(
- false, &fake_clock_, &transport, nullptr, nullptr, nullptr, nullptr,
+ false, &fake_clock_, &transport, nullptr, absl::nullopt, nullptr, nullptr,
nullptr, nullptr, &mock_rtc_event_log_, nullptr,
&retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
rtp_sender_->SetSequenceNumber(kSeqNum);
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.cc b/modules/rtp_rtcp/source/rtp_sender_video.cc
index 949af14..c63f0d7 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -21,6 +21,7 @@
#include "absl/memory/memory.h"
#include "absl/strings/match.h"
#include "api/crypto/frame_encryptor_interface.h"
+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
@@ -152,6 +153,20 @@
return true;
}
+const char* FrameTypeToString(FrameType frame_type) {
+ switch (frame_type) {
+ case kEmptyFrame:
+ return "empty";
+ case kVideoFrameKey:
+ return "video_key";
+ case kVideoFrameDelta:
+ return "video_delta";
+ default:
+ RTC_NOTREACHED();
+ return "";
+ }
+}
+
} // namespace
RTPSenderVideo::RTPSenderVideo(Clock* clock,
@@ -207,8 +222,8 @@
// Remember some values about the packet before sending it away.
size_t packet_size = packet->size();
uint16_t seq_num = packet->SequenceNumber();
- if (!rtp_sender_->SendToNetwork(std::move(packet), storage,
- RtpPacketSender::kLowPriority)) {
+ if (!LogAndSendToNetwork(std::move(packet), storage,
+ RtpPacketSender::kLowPriority)) {
RTC_LOG(LS_WARNING) << "Failed to send video packet " << seq_num;
return;
}
@@ -249,8 +264,8 @@
}
// Send |red_packet| instead of |packet| for allocated sequence number.
size_t red_packet_size = red_packet->size();
- if (rtp_sender_->SendToNetwork(std::move(red_packet), media_packet_storage,
- RtpPacketSender::kLowPriority)) {
+ if (LogAndSendToNetwork(std::move(red_packet), media_packet_storage,
+ RtpPacketSender::kLowPriority)) {
rtc::CritScope cs(&stats_crit_);
video_bitrate_.Update(red_packet_size, clock_->TimeInMilliseconds());
} else {
@@ -265,8 +280,8 @@
rtp_packet->set_capture_time_ms(media_packet->capture_time_ms());
rtp_packet->set_is_fec(true);
uint16_t fec_sequence_number = rtp_packet->SequenceNumber();
- if (rtp_sender_->SendToNetwork(std::move(rtp_packet), kDontRetransmit,
- RtpPacketSender::kLowPriority)) {
+ if (LogAndSendToNetwork(std::move(rtp_packet), kDontRetransmit,
+ RtpPacketSender::kLowPriority)) {
rtc::CritScope cs(&stats_crit_);
fec_bitrate_.Update(fec_packet->length(), clock_->TimeInMilliseconds());
} else {
@@ -293,8 +308,8 @@
for (auto& fec_packet : fec_packets) {
size_t packet_length = fec_packet->size();
uint16_t seq_num = fec_packet->SequenceNumber();
- if (rtp_sender_->SendToNetwork(std::move(fec_packet), kDontRetransmit,
- RtpPacketSender::kLowPriority)) {
+ if (LogAndSendToNetwork(std::move(fec_packet), kDontRetransmit,
+ RtpPacketSender::kLowPriority)) {
rtc::CritScope cs(&stats_crit_);
fec_bitrate_.Update(packet_length, clock_->TimeInMilliseconds());
} else {
@@ -304,6 +319,24 @@
}
}
+bool RTPSenderVideo::LogAndSendToNetwork(
+ std::unique_ptr<RtpPacketToSend> packet,
+ StorageType storage,
+ RtpPacketSender::Priority priority) {
+#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
+ int64_t now_ms = clock_->TimeInMilliseconds();
+ BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoTotBitrate_kbps", now_ms,
+ rtp_sender_->ActualSendBitrateKbit(),
+ packet->Ssrc());
+ BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoFecBitrate_kbps", now_ms,
+ FecOverheadRate() / 1000, packet->Ssrc());
+ BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoNackBitrate_kbps", now_ms,
+ rtp_sender_->NackOverheadRate() / 1000,
+ packet->Ssrc());
+#endif
+ return rtp_sender_->SendToNetwork(std::move(packet), storage, priority);
+}
+
void RTPSenderVideo::SetUlpfecConfig(int red_payload_type,
int ulpfec_payload_type) {
// Sanity check. Per the definition of UlpfecConfig (see config.h),
@@ -371,6 +404,15 @@
const RTPFragmentationHeader* fragmentation,
const RTPVideoHeader* video_header,
int64_t expected_retransmission_time_ms) {
+ RTC_DCHECK(frame_type == kVideoFrameKey || frame_type == kVideoFrameDelta ||
+ frame_type == kEmptyFrame);
+
+ TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", capture_time_ms, "Send", "type",
+ FrameTypeToString(frame_type));
+
+ if (frame_type == kEmptyFrame)
+ return true;
+
if (payload_size == 0)
return false;
RTC_CHECK(video_header);
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.h b/modules/rtp_rtcp/source/rtp_sender_video.h
index 8e26206..d29934f 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.h
+++ b/modules/rtp_rtcp/source/rtp_sender_video.h
@@ -121,6 +121,10 @@
StorageType media_packet_storage,
bool protect_media_packet);
+ bool LogAndSendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
+ StorageType storage,
+ RtpPacketSender::Priority priority);
+
bool red_enabled() const RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_) {
return red_payload_type_ >= 0;
}
diff --git a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
index 9b155e7..85bbc80 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
@@ -112,7 +112,7 @@
&fake_clock_,
&transport_,
nullptr,
- nullptr,
+ absl::nullopt,
nullptr,
nullptr,
nullptr,
