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Pass Clock and TaskQueueFactory as Environment in voip audio channel

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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}
This commit is contained in:
Danil Chapovalov 2024-08-02 11:31:59 +02:00 committed by WebRTC LUCI CQ
parent b4462510c3
commit 943828b7ff
11 changed files with 30 additions and 40 deletions
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2
audio/voip/BUILD.gn
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@ -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",

13
audio/voip/audio_channel.cc
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@ -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(clock);
receive_statistics_ = ReceiveStatistics::Create(&env.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_ =
std::make_unique<AudioEgress>(rtp_rtcp_.get(), clock, task_queue_factory);
egress_ = std::make_unique<AudioEgress>(env, rtp_rtcp_.get());
// Set the instance of audio ingress to be part of audio mixer for ADM to
// fetch audio samples to play.

4
audio/voip/audio_channel.h
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@ -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;

8
audio/voip/audio_egress.cc
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@ -18,13 +18,11 @@
namespace webrtc {
AudioEgress::AudioEgress(RtpRtcpInterface* rtp_rtcp,
Clock* clock,
TaskQueueFactory* task_queue_factory)
AudioEgress::AudioEgress(const Environment& env, RtpRtcpInterface* rtp_rtcp)
: 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()) {

5
audio/voip/audio_egress.h
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@ -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,
Clock* clock,
TaskQueueFactory* task_queue_factory);
AudioEgress(const Environment& env, RtpRtcpInterface* rtp_rtcp);
~AudioEgress() override;
// Set the encoder format and payload type for AudioCodingModule.

12
audio/voip/audio_ingress.cc
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@ -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),
clock_(clock) {}
ntp_estimator_(&env_.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()) !=
0) {
if (acm_receiver_.InsertPacket(header, data_view,
env_.clock().CurrentTime()) != 0) {
RTC_DLOG(LS_ERROR) << "AudioIngress::ReceivedRTPPacket() unable to "
"push data to the ACM";
}

9
audio/voip/audio_ingress.h
View File

@ -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,
Clock* clock,
AudioIngress(const Environment& env,
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_);

3
audio/voip/test/audio_channel_unittest.cc
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@ -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(),
decoder_factory_);
env_, &transport_, ssrc, audio_mixer_.get(), decoder_factory_);
audio_channel->SetEncoder(
kPcmuPayload, kPcmuFormat,
encoder_factory_->Create(env_, kPcmuFormat,

4
audio/voip/test/audio_egress_unittest.cc
View File

@ -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>(
rtp_rtcp_.get(), time_controller_.GetClock(),
time_controller_.GetTaskQueueFactory());
egress_ = std::make_unique<AudioEgress>(env_, rtp_rtcp_.get());
constexpr int kPcmuPayload = 0;
egress_->SetEncoder(kPcmuPayload, kPcmuFormat,
encoder_factory_->Create(

7
audio/voip/test/audio_ingress_unittest.cc
View File

@ -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(),
decoder_factory_);
env_, rtp_rtcp_.get(), receive_statistics_.get(), decoder_factory_);
ingress_->SetReceiveCodecs({{kPcmuPayload, kPcmuFormat}});
egress_ = std::make_unique<AudioEgress>(
rtp_rtcp_.get(), time_controller_.GetClock(),
time_controller_.GetTaskQueueFactory());
egress_ = std::make_unique<AudioEgress>(env_, rtp_rtcp_.get());
egress_->SetEncoder(kPcmuPayload, kPcmuFormat,
encoder_factory_->Create(
env_, kPcmuFormat, {.payload_type = kPcmuPayload}));

3
audio/voip/voip_core.cc
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@ -132,8 +132,7 @@ ChannelId VoipCore::CreateChannel(Transport* transport,
}
rtc::scoped_refptr<AudioChannel> channel =
rtc::make_ref_counted<AudioChannel>(transport, local_ssrc.value(),
&env_.task_queue_factory(),
rtc::make_ref_counted<AudioChannel>(env_, transport, local_ssrc.value(),
audio_mixer_.get(), decoder_factory_);
{