Injecting Clock into audio streams.
Bug: webrtc:10365
Change-Id: Ia47fd806b84d94fd90b734c87c5e338e36fb695a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125191
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26969}diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index a6c81fc..47530c3 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -67,6 +67,7 @@
namespace internal {
namespace {
std::unique_ptr<voe::ChannelReceiveInterface> CreateChannelReceive(
+ Clock* clock,
webrtc::AudioState* audio_state,
ProcessThread* module_process_thread,
const webrtc::AudioReceiveStream::Config& config,
@@ -75,7 +76,7 @@
internal::AudioState* internal_audio_state =
static_cast<internal::AudioState*>(audio_state);
return voe::CreateChannelReceive(
- module_process_thread, internal_audio_state->audio_device_module(),
+ clock, module_process_thread, internal_audio_state->audio_device_module(),
config.media_transport, config.rtcp_send_transport, event_log,
config.rtp.remote_ssrc, config.jitter_buffer_max_packets,
config.jitter_buffer_fast_accelerate, config.jitter_buffer_min_delay_ms,
@@ -85,23 +86,27 @@
} // namespace
AudioReceiveStream::AudioReceiveStream(
+ Clock* clock,
RtpStreamReceiverControllerInterface* receiver_controller,
PacketRouter* packet_router,
ProcessThread* module_process_thread,
const webrtc::AudioReceiveStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
webrtc::RtcEventLog* event_log)
- : AudioReceiveStream(receiver_controller,
+ : AudioReceiveStream(clock,
+ receiver_controller,
packet_router,
config,
audio_state,
event_log,
- CreateChannelReceive(audio_state.get(),
+ CreateChannelReceive(clock,
+ audio_state.get(),
module_process_thread,
config,
event_log)) {}
AudioReceiveStream::AudioReceiveStream(
+ Clock* clock,
RtpStreamReceiverControllerInterface* receiver_controller,
PacketRouter* packet_router,
const webrtc::AudioReceiveStream::Config& config,
diff --git a/audio/audio_receive_stream.h b/audio/audio_receive_stream.h
index b6d0aa5..1745c0f 100644
--- a/audio/audio_receive_stream.h
+++ b/audio/audio_receive_stream.h
@@ -21,6 +21,7 @@
#include "call/syncable.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/thread_checker.h"
+#include "system_wrappers/include/clock.h"
namespace webrtc {
class PacketRouter;
@@ -41,7 +42,8 @@
public AudioMixer::Source,
public Syncable {
public:
- AudioReceiveStream(RtpStreamReceiverControllerInterface* receiver_controller,
+ AudioReceiveStream(Clock* clock,
+ RtpStreamReceiverControllerInterface* receiver_controller,
PacketRouter* packet_router,
ProcessThread* module_process_thread,
const webrtc::AudioReceiveStream::Config& config,
@@ -49,6 +51,7 @@
webrtc::RtcEventLog* event_log);
// For unit tests, which need to supply a mock channel receive.
AudioReceiveStream(
+ Clock* clock,
RtpStreamReceiverControllerInterface* receiver_controller,
PacketRouter* packet_router,
const webrtc::AudioReceiveStream::Config& config,
diff --git a/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc
index b64626f..4594e56 100644
--- a/audio/audio_receive_stream_unittest.cc
+++ b/audio/audio_receive_stream_unittest.cc
@@ -115,8 +115,8 @@
std::unique_ptr<internal::AudioReceiveStream> CreateAudioReceiveStream() {
return std::unique_ptr<internal::AudioReceiveStream>(
new internal::AudioReceiveStream(
- &rtp_stream_receiver_controller_, &packet_router_, stream_config_,
- audio_state_, &event_log_,
+ Clock::GetRealTimeClock(), &rtp_stream_receiver_controller_,
+ &packet_router_, stream_config_, audio_state_, &event_log_,
std::unique_ptr<voe::ChannelReceiveInterface>(channel_receive_)));
}
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 2e4830c..bb79149 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -37,7 +37,6 @@
#include "rtc_base/logging.h"
#include "rtc_base/strings/audio_format_to_string.h"
#include "rtc_base/task_queue.h"
-#include "rtc_base/time_utils.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
@@ -83,6 +82,7 @@
} // namespace
AudioSendStream::AudioSendStream(
+ Clock* clock,
const webrtc::AudioSendStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
rtc::TaskQueue* worker_queue,
@@ -92,7 +92,8 @@
RtcEventLog* event_log,
RtcpRttStats* rtcp_rtt_stats,
const absl::optional<RtpState>& suspended_rtp_state)
- : AudioSendStream(config,
+ : AudioSendStream(clock,
+ config,
audio_state,
worker_queue,
rtp_transport,
@@ -100,7 +101,8 @@
event_log,
rtcp_rtt_stats,
suspended_rtp_state,
- voe::CreateChannelSend(worker_queue,
+ voe::CreateChannelSend(clock,
+ worker_queue,
module_process_thread,
config.media_transport,
/*overhead_observer=*/this,
@@ -113,6 +115,7 @@
config.rtcp_report_interval_ms)) {}
AudioSendStream::AudioSendStream(
+ Clock* clock,
const webrtc::AudioSendStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
rtc::TaskQueue* worker_queue,
@@ -122,7 +125,8 @@
RtcpRttStats* rtcp_rtt_stats,
const absl::optional<RtpState>& suspended_rtp_state,
std::unique_ptr<voe::ChannelSendInterface> channel_send)
- : worker_queue_(worker_queue),
+ : clock_(clock),
+ worker_queue_(worker_queue),
config_(Config(/*send_transport=*/nullptr,
/*media_transport=*/nullptr)),
audio_state_(audio_state),
@@ -455,7 +459,7 @@
// TODO(eladalon): This function call could potentially reset the window,
// setting both PLR and RPLR to unknown. Consider (during upcoming
// refactoring) passing an indication of such an event.
- packet_loss_tracker_.OnPacketAdded(seq_num, rtc::TimeMillis());
+ packet_loss_tracker_.OnPacketAdded(seq_num, clock_->TimeInMilliseconds());
}
}
diff --git a/audio/audio_send_stream.h b/audio/audio_send_stream.h
index a1dab79..c227e61 100644
--- a/audio/audio_send_stream.h
+++ b/audio/audio_send_stream.h
@@ -39,7 +39,8 @@
public webrtc::PacketFeedbackObserver,
public webrtc::OverheadObserver {
public:
- AudioSendStream(const webrtc::AudioSendStream::Config& config,
+ AudioSendStream(Clock* clock,
+ const webrtc::AudioSendStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
rtc::TaskQueue* worker_queue,
ProcessThread* module_process_thread,
@@ -49,7 +50,8 @@
RtcpRttStats* rtcp_rtt_stats,
const absl::optional<RtpState>& suspended_rtp_state);
// For unit tests, which need to supply a mock ChannelSend.
- AudioSendStream(const webrtc::AudioSendStream::Config& config,
+ AudioSendStream(Clock* clock,
+ const webrtc::AudioSendStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
rtc::TaskQueue* worker_queue,
RtpTransportControllerSendInterface* rtp_transport,
@@ -135,6 +137,7 @@
RTC_EXCLUSIVE_LOCKS_REQUIRED(overhead_per_packet_lock_);
void RegisterCngPayloadType(int payload_type, int clockrate_hz);
+ Clock* clock_;
rtc::ThreadChecker worker_thread_checker_;
rtc::ThreadChecker pacer_thread_checker_;
diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 21f2218..ddd8137 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -163,8 +163,9 @@
std::unique_ptr<internal::AudioSendStream> CreateAudioSendStream() {
return std::unique_ptr<internal::AudioSendStream>(
new internal::AudioSendStream(
- stream_config_, audio_state_, &worker_queue_, &rtp_transport_,
- &bitrate_allocator_, &event_log_, &rtcp_rtt_stats_, absl::nullopt,
+ Clock::GetRealTimeClock(), stream_config_, audio_state_,
+ &worker_queue_, &rtp_transport_, &bitrate_allocator_, &event_log_,
+ &rtcp_rtt_stats_, absl::nullopt,
std::unique_ptr<voe::ChannelSendInterface>(channel_send_)));
}
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 0e218ed..79523c8 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -76,7 +76,8 @@
public MediaTransportAudioSinkInterface {
public:
// Used for receive streams.
- ChannelReceive(ProcessThread* module_process_thread,
+ ChannelReceive(Clock* clock,
+ ProcessThread* module_process_thread,
AudioDeviceModule* audio_device_module,
MediaTransportInterface* media_transport,
Transport* rtcp_send_transport,
@@ -428,6 +429,7 @@
}
ChannelReceive::ChannelReceive(
+ Clock* clock,
ProcessThread* module_process_thread,
AudioDeviceModule* audio_device_module,
MediaTransportInterface* media_transport,
@@ -443,11 +445,10 @@
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options)
: event_log_(rtc_event_log),
- rtp_receive_statistics_(
- ReceiveStatistics::Create(Clock::GetRealTimeClock())),
+ rtp_receive_statistics_(ReceiveStatistics::Create(clock)),
remote_ssrc_(remote_ssrc),
_outputAudioLevel(),
- ntp_estimator_(Clock::GetRealTimeClock()),
+ ntp_estimator_(clock),
playout_timestamp_rtp_(0),
playout_delay_ms_(0),
rtp_ts_wraparound_handler_(new rtc::TimestampWrapAroundHandler()),
@@ -480,6 +481,7 @@
rtp_receive_statistics_->EnableRetransmitDetection(remote_ssrc_, true);
RtpRtcp::Configuration configuration;
+ configuration.clock = clock;
configuration.audio = true;
configuration.receiver_only = true;
configuration.outgoing_transport = rtcp_send_transport;
@@ -959,6 +961,7 @@
} // namespace
std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
+ Clock* clock,
ProcessThread* module_process_thread,
AudioDeviceModule* audio_device_module,
MediaTransportInterface* media_transport,
@@ -974,7 +977,7 @@
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options) {
return absl::make_unique<ChannelReceive>(
- module_process_thread, audio_device_module, media_transport,
+ clock, module_process_thread, audio_device_module, media_transport,
rtcp_send_transport, rtc_event_log, remote_ssrc,
jitter_buffer_max_packets, jitter_buffer_fast_playout,
jitter_buffer_min_delay_ms, jitter_buffer_enable_rtx_handling,
diff --git a/audio/channel_receive.h b/audio/channel_receive.h
index 28c7c06..b3e4f9c 100644
--- a/audio/channel_receive.h
+++ b/audio/channel_receive.h
@@ -136,6 +136,7 @@
};
std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
+ Clock* clock,
ProcessThread* module_process_thread,
AudioDeviceModule* audio_device_module,
MediaTransportInterface* media_transport,
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 2db4ff7..813795b 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -41,6 +41,7 @@
#include "rtc_base/task_queue.h"
#include "rtc_base/thread_checker.h"
#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"
@@ -85,7 +86,8 @@
// declaration.
friend class VoERtcpObserver;
- ChannelSend(rtc::TaskQueue* encoder_queue,
+ ChannelSend(Clock* clock,
+ rtc::TaskQueue* encoder_queue,
ProcessThread* module_process_thread,
MediaTransportInterface* media_transport,
OverheadObserver* overhead_observer,
@@ -606,7 +608,8 @@
return 0;
}
-ChannelSend::ChannelSend(rtc::TaskQueue* encoder_queue,
+ChannelSend::ChannelSend(Clock* clock,
+ rtc::TaskQueue* encoder_queue,
ProcessThread* module_process_thread,
MediaTransportInterface* media_transport,
OverheadObserver* overhead_observer,
@@ -628,8 +631,8 @@
feedback_observer_proxy_(new TransportFeedbackProxy()),
seq_num_allocator_proxy_(new TransportSequenceNumberProxy()),
rtp_packet_sender_proxy_(new RtpPacketSenderProxy()),
- retransmission_rate_limiter_(new RateLimiter(Clock::GetRealTimeClock(),
- kMaxRetransmissionWindowMs)),
+ retransmission_rate_limiter_(
+ new RateLimiter(clock, kMaxRetransmissionWindowMs)),
use_twcc_plr_for_ana_(
webrtc::field_trial::FindFullName("UseTwccPlrForAna") == "Enabled"),
encoder_queue_(encoder_queue),
@@ -659,6 +662,7 @@
configuration.transport_feedback_callback = feedback_observer_proxy_.get();
}
+ configuration.clock = clock;
configuration.audio = true;
configuration.outgoing_transport = rtp_transport;
@@ -1217,6 +1221,7 @@
} // namespace
std::unique_ptr<ChannelSendInterface> CreateChannelSend(
+ Clock* clock,
rtc::TaskQueue* encoder_queue,
ProcessThread* module_process_thread,
MediaTransportInterface* media_transport,
@@ -1229,9 +1234,10 @@
bool extmap_allow_mixed,
int rtcp_report_interval_ms) {
return absl::make_unique<ChannelSend>(
- encoder_queue, module_process_thread, media_transport, overhead_observer,
- rtp_transport, rtcp_rtt_stats, rtc_event_log, frame_encryptor,
- crypto_options, extmap_allow_mixed, rtcp_report_interval_ms);
+ clock, encoder_queue, module_process_thread, media_transport,
+ overhead_observer, rtp_transport, rtcp_rtt_stats, rtc_event_log,
+ frame_encryptor, crypto_options, extmap_allow_mixed,
+ rtcp_report_interval_ms);
}
} // namespace voe
diff --git a/audio/channel_send.h b/audio/channel_send.h
index 5f3dd0f..4ab53c0 100644
--- a/audio/channel_send.h
+++ b/audio/channel_send.h
@@ -115,6 +115,7 @@
};
std::unique_ptr<ChannelSendInterface> CreateChannelSend(
+ Clock* clock,
rtc::TaskQueue* encoder_queue,
ProcessThread* module_process_thread,
MediaTransportInterface* media_transport,
diff --git a/audio/test/media_transport_test.cc b/audio/test/media_transport_test.cc
index 3fd451b..02e3f75 100644
--- a/audio/test/media_transport_test.cc
+++ b/audio/test/media_transport_test.cc
@@ -106,6 +106,7 @@
ProcessThread::Create("audio recv thread");
webrtc::internal::AudioReceiveStream receive_stream(
+ Clock::GetRealTimeClock(),
/*rtp_stream_receiver_controller=*/nullptr,
/*packet_router=*/nullptr, receive_process_thread.get(), receive_config,
audio_state, null_event_log.get());
@@ -120,7 +121,8 @@
std::unique_ptr<ProcessThread> send_process_thread =
ProcessThread::Create("audio send thread");
webrtc::internal::AudioSendStream send_stream(
- send_config, audio_state, &send_tq, send_process_thread.get(),
+ Clock::GetRealTimeClock(), send_config, audio_state, &send_tq,
+ send_process_thread.get(),
/*transport=*/nullptr, &bitrate_allocator, null_event_log.get(),
/*rtcp_rtt_stats=*/nullptr, absl::optional<RtpState>());
diff --git a/call/call.cc b/call/call.cc
index 7c80779..98747cf 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -670,10 +670,10 @@
// having it injected.
AudioSendStream* send_stream = new AudioSendStream(
- config, config_.audio_state, transport_send_ptr_->GetWorkerQueue(),
- module_process_thread_.get(), transport_send_ptr_,
- bitrate_allocator_.get(), event_log_, call_stats_.get(),
- suspended_rtp_state);
+ clock_, config, config_.audio_state,
+ transport_send_ptr_->GetWorkerQueue(), module_process_thread_.get(),
+ transport_send_ptr_, bitrate_allocator_.get(), event_log_,
+ call_stats_.get(), suspended_rtp_state);
{
WriteLockScoped write_lock(*send_crit_);
RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) ==
@@ -729,7 +729,7 @@
event_log_->Log(absl::make_unique<RtcEventAudioReceiveStreamConfig>(
CreateRtcLogStreamConfig(config)));
AudioReceiveStream* receive_stream = new AudioReceiveStream(
- &audio_receiver_controller_, transport_send_ptr_->packet_router(),
+ clock_, &audio_receiver_controller_, transport_send_ptr_->packet_router(),
module_process_thread_.get(), config, config_.audio_state, event_log_);
{
WriteLockScoped write_lock(*receive_crit_);
