Pass Clock and TaskQueueFactory as Environment in voip audio channel

To make Environment available for creating AudioDecoders to use propagated field trials

Bug: webrtc:356878416
Change-Id: Ic2371f038b75402bbd007c948f43c60cc6cca8a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358400
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42718}

audio/voip/BUILD.gn

@@ -63,6 +63,7 @@ rtc_library("audio_ingress") {
     "../../api:transport_api",
     "../../api/audio:audio_mixer_api",
     "../../api/audio_codecs:audio_codecs_api",
+    "../../api/environment",
     "../../api/voip:voip_api",
     "../../modules/audio_coding",
     "../../modules/rtp_rtcp",
@@ -87,6 +88,7 @@ rtc_library("audio_egress") {
     "..:audio",
     "../../api:sequence_checker",
     "../../api/audio_codecs:audio_codecs_api",
+    "../../api/environment",
     "../../api/task_queue",
     "../../call:audio_sender_interface",
     "../../modules/audio_coding",

audio/voip/audio_channel.cc

@@ -28,20 +28,18 @@ constexpr int kRtcpReportIntervalMs = 5000;
 }  // namespace
 
 AudioChannel::AudioChannel(
+    const Environment& env,
     Transport* transport,
     uint32_t local_ssrc,
-    TaskQueueFactory* task_queue_factory,
     AudioMixer* audio_mixer,
     rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
     : audio_mixer_(audio_mixer) {
-  RTC_DCHECK(task_queue_factory);
   RTC_DCHECK(audio_mixer);
 
-  Clock* clock = Clock::GetRealTimeClock();
+  receive_statistics_ = ReceiveStatistics::Create(&env.clock());
-  receive_statistics_ = ReceiveStatistics::Create(clock);
 
   RtpRtcpInterface::Configuration rtp_config;
-  rtp_config.clock = clock;
+  rtp_config.clock = &env.clock();
   rtp_config.audio = true;
   rtp_config.receive_statistics = receive_statistics_.get();
   rtp_config.rtcp_report_interval_ms = kRtcpReportIntervalMs;
@@ -53,11 +51,10 @@ AudioChannel::AudioChannel(
   rtp_rtcp_->SetSendingMediaStatus(false);
   rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound);
 
-  ingress_ = std::make_unique<AudioIngress>(rtp_rtcp_.get(), clock,
+  ingress_ = std::make_unique<AudioIngress>(env, rtp_rtcp_.get(),
                                             receive_statistics_.get(),
                                             std::move(decoder_factory));
-  egress_ =
+  egress_ = std::make_unique<AudioEgress>(env, rtp_rtcp_.get());
-      std::make_unique<AudioEgress>(rtp_rtcp_.get(), clock, task_queue_factory);
 
   // Set the instance of audio ingress to be part of audio mixer for ADM to
   // fetch audio samples to play.

audio/voip/audio_channel.h

@@ -31,9 +31,9 @@ namespace webrtc {
 // these two classes as it has both sending and receiving capabilities.
 class AudioChannel : public RefCountInterface {
  public:
-  AudioChannel(Transport* transport,
+  AudioChannel(const Environment& env,
+               Transport* transport,
                uint32_t local_ssrc,
-               TaskQueueFactory* task_queue_factory,
                AudioMixer* audio_mixer,
                rtc::scoped_refptr<AudioDecoderFactory> decoder_factory);
   ~AudioChannel() override;

audio/voip/audio_egress.cc

@@ -18,13 +18,11 @@
 
 namespace webrtc {
 
-AudioEgress::AudioEgress(RtpRtcpInterface* rtp_rtcp,
+AudioEgress::AudioEgress(const Environment& env, RtpRtcpInterface* rtp_rtcp)
-                         Clock* clock,
-                         TaskQueueFactory* task_queue_factory)
     : rtp_rtcp_(rtp_rtcp),
-      rtp_sender_audio_(clock, rtp_rtcp_->RtpSender()),
+      rtp_sender_audio_(&env.clock(), rtp_rtcp_->RtpSender()),
       audio_coding_(AudioCodingModule::Create()),
-      encoder_queue_(task_queue_factory->CreateTaskQueue(
+      encoder_queue_(env.task_queue_factory().CreateTaskQueue(
           "AudioEncoder",
           TaskQueueFactory::Priority::NORMAL)),
       encoder_queue_checker_(encoder_queue_.get()) {

audio/voip/audio_egress.h

@@ -15,6 +15,7 @@
 #include <string>
 
 #include "api/audio_codecs/audio_format.h"
+#include "api/environment/environment.h"
 #include "api/sequence_checker.h"
 #include "api/task_queue/task_queue_base.h"
 #include "api/task_queue/task_queue_factory.h"
@@ -46,9 +47,7 @@ namespace webrtc {
 // smaller footprint.
 class AudioEgress : public AudioSender, public AudioPacketizationCallback {
  public:
-  AudioEgress(RtpRtcpInterface* rtp_rtcp,
+  AudioEgress(const Environment& env, RtpRtcpInterface* rtp_rtcp);
-              Clock* clock,
-              TaskQueueFactory* task_queue_factory);
   ~AudioEgress() override;
 
   // Set the encoder format and payload type for AudioCodingModule.

audio/voip/audio_ingress.cc

@@ -41,18 +41,18 @@ acm2::AcmReceiver::Config CreateAcmConfig(
 }  // namespace
 
 AudioIngress::AudioIngress(
+    const Environment& env,
     RtpRtcpInterface* rtp_rtcp,
-    Clock* clock,
     ReceiveStatistics* receive_statistics,
     rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
-    : playing_(false),
+    : env_(env),
+      playing_(false),
       remote_ssrc_(0),
       first_rtp_timestamp_(-1),
       rtp_receive_statistics_(receive_statistics),
       rtp_rtcp_(rtp_rtcp),
       acm_receiver_(CreateAcmConfig(decoder_factory)),
-      ntp_estimator_(clock),
+      ntp_estimator_(&env_.clock()) {}
-      clock_(clock) {}
 
 AudioIngress::~AudioIngress() = default;
 
@@ -186,8 +186,8 @@ void AudioIngress::ReceivedRTPPacket(rtc::ArrayView<const uint8_t> rtp_packet) {
   auto data_view = rtc::ArrayView<const uint8_t>(payload, payload_data_length);
 
   // Push the incoming payload (parsed and ready for decoding) into the ACM.
-  if (acm_receiver_.InsertPacket(header, data_view, clock_->CurrentTime()) !=
+  if (acm_receiver_.InsertPacket(header, data_view,
-      0) {
+                                 env_.clock().CurrentTime()) != 0) {
     RTC_DLOG(LS_ERROR) << "AudioIngress::ReceivedRTPPacket() unable to "
                           "push data to the ACM";
   }

audio/voip/audio_ingress.h

@@ -20,6 +20,7 @@
 #include "absl/types/optional.h"
 #include "api/array_view.h"
 #include "api/audio/audio_mixer.h"
+#include "api/environment/environment.h"
 #include "api/rtp_headers.h"
 #include "api/scoped_refptr.h"
 #include "api/voip/voip_statistics.h"
@@ -46,8 +47,8 @@ namespace webrtc {
 // smaller footprint.
 class AudioIngress : public AudioMixer::Source {
  public:
-  AudioIngress(RtpRtcpInterface* rtp_rtcp,
+  AudioIngress(const Environment& env,
-               Clock* clock,
+               RtpRtcpInterface* rtp_rtcp,
                ReceiveStatistics* receive_statistics,
                rtc::scoped_refptr<AudioDecoderFactory> decoder_factory);
   ~AudioIngress() override;
@@ -105,6 +106,8 @@ class AudioIngress : public AudioMixer::Source {
   }
 
  private:
+  const Environment env_;
+
   // Indicates AudioIngress status as caller invokes Start/StopPlaying.
   // If not playing, incoming RTP data processing is skipped, thus
   // producing no data to output device.
@@ -133,8 +136,6 @@ class AudioIngress : public AudioMixer::Source {
 
   RemoteNtpTimeEstimator ntp_estimator_ RTC_GUARDED_BY(lock_);
 
-  Clock* clock_;
-
   // For receiving RTP statistics, this tracks the sampling rate value
   // per payload type set when caller set via SetReceiveCodecs.
   std::map<int, int> receive_codec_info_ RTC_GUARDED_BY(lock_);

audio/voip/test/audio_channel_unittest.cc

@@ -73,8 +73,7 @@ class AudioChannelTest : public ::testing::Test {
     // simplify network routing logic.
     rtc::scoped_refptr<AudioChannel> audio_channel =
         rtc::make_ref_counted<AudioChannel>(
-            &transport_, ssrc, &env_.task_queue_factory(), audio_mixer_.get(),
+            env_, &transport_, ssrc, audio_mixer_.get(), decoder_factory_);
-            decoder_factory_);
     audio_channel->SetEncoder(
         kPcmuPayload, kPcmuFormat,
         encoder_factory_->Create(env_, kPcmuFormat,

audio/voip/test/audio_egress_unittest.cc

@@ -70,9 +70,7 @@ class AudioEgressTest : public ::testing::Test {
   void SetUp() override {
     rtp_rtcp_ =
         CreateRtpStack(time_controller_.GetClock(), &transport_, kRemoteSsrc);
-    egress_ = std::make_unique<AudioEgress>(
+    egress_ = std::make_unique<AudioEgress>(env_, rtp_rtcp_.get());
-        rtp_rtcp_.get(), time_controller_.GetClock(),
-        time_controller_.GetTaskQueueFactory());
     constexpr int kPcmuPayload = 0;
     egress_->SetEncoder(kPcmuPayload, kPcmuFormat,
                         encoder_factory_->Create(

audio/voip/test/audio_ingress_unittest.cc

@@ -64,13 +64,10 @@ class AudioIngressTest : public ::testing::Test {
   void SetUp() override {
     constexpr int kPcmuPayload = 0;
     ingress_ = std::make_unique<AudioIngress>(
-        rtp_rtcp_.get(), time_controller_.GetClock(), receive_statistics_.get(),
+        env_, rtp_rtcp_.get(), receive_statistics_.get(), decoder_factory_);
-        decoder_factory_);
     ingress_->SetReceiveCodecs({{kPcmuPayload, kPcmuFormat}});
 
-    egress_ = std::make_unique<AudioEgress>(
+    egress_ = std::make_unique<AudioEgress>(env_, rtp_rtcp_.get());
-        rtp_rtcp_.get(), time_controller_.GetClock(),
-        time_controller_.GetTaskQueueFactory());
     egress_->SetEncoder(kPcmuPayload, kPcmuFormat,
                         encoder_factory_->Create(
                             env_, kPcmuFormat, {.payload_type = kPcmuPayload}));

audio/voip/voip_core.cc

@@ -132,8 +132,7 @@ ChannelId VoipCore::CreateChannel(Transport* transport,
   }
 
   rtc::scoped_refptr<AudioChannel> channel =
-      rtc::make_ref_counted<AudioChannel>(transport, local_ssrc.value(),
+      rtc::make_ref_counted<AudioChannel>(env_, transport, local_ssrc.value(),
-                                          &env_.task_queue_factory(),
                                           audio_mixer_.get(), decoder_factory_);
 
   {