Delete class ChannelReceiveProxy.
Replaced by an interface ChannelReceiveInterface, implemented
by ChannelReceive and the corresponding mock class.
Moved thread checkers to ChannelReceive. That class is moved to the
anonymous namespace in the .cc file, and exposed only via a function
CreateChannelReceive.
Bug: webrtc:9801
Change-Id: Iecacbb1858885bf86da9484f2422e53323dbe87a
Reviewed-on: https://webrtc-review.googlesource.com/c/110610
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25665}diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index 4f2e29c..4848491 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -21,7 +21,6 @@
#include "audio/audio_send_stream.h"
#include "audio/audio_state.h"
#include "audio/channel_receive.h"
-#include "audio/channel_receive_proxy.h"
#include "audio/conversion.h"
#include "call/rtp_config.h"
#include "call/rtp_stream_receiver_controller_interface.h"
@@ -68,7 +67,7 @@
namespace internal {
namespace {
-std::unique_ptr<voe::ChannelReceiveProxy> CreateChannelAndProxy(
+std::unique_ptr<voe::ChannelReceiveInterface> CreateChannelReceive(
webrtc::AudioState* audio_state,
ProcessThread* module_process_thread,
const webrtc::AudioReceiveStream::Config& config,
@@ -76,13 +75,12 @@
RTC_DCHECK(audio_state);
internal::AudioState* internal_audio_state =
static_cast<internal::AudioState*>(audio_state);
- return absl::make_unique<voe::ChannelReceiveProxy>(
- absl::make_unique<voe::ChannelReceive>(
- 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.decoder_factory,
- config.codec_pair_id, config.frame_decryptor, config.crypto_options));
+ return voe::CreateChannelReceive(
+ 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.decoder_factory,
+ config.codec_pair_id, config.frame_decryptor, config.crypto_options);
}
} // namespace
@@ -98,10 +96,10 @@
config,
audio_state,
event_log,
- CreateChannelAndProxy(audio_state.get(),
- module_process_thread,
- config,
- event_log)) {}
+ CreateChannelReceive(audio_state.get(),
+ module_process_thread,
+ config,
+ event_log)) {}
AudioReceiveStream::AudioReceiveStream(
RtpStreamReceiverControllerInterface* receiver_controller,
@@ -109,13 +107,13 @@
const webrtc::AudioReceiveStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
webrtc::RtcEventLog* event_log,
- std::unique_ptr<voe::ChannelReceiveProxy> channel_proxy)
- : audio_state_(audio_state), channel_proxy_(std::move(channel_proxy)) {
+ std::unique_ptr<voe::ChannelReceiveInterface> channel_receive)
+ : audio_state_(audio_state), channel_receive_(std::move(channel_receive)) {
RTC_LOG(LS_INFO) << "AudioReceiveStream: " << config.rtp.remote_ssrc;
RTC_DCHECK(config.decoder_factory);
RTC_DCHECK(config.rtcp_send_transport);
RTC_DCHECK(audio_state_);
- RTC_DCHECK(channel_proxy_);
+ RTC_DCHECK(channel_receive_);
module_process_thread_checker_.DetachFromThread();
@@ -123,11 +121,11 @@
RTC_DCHECK(receiver_controller);
RTC_DCHECK(packet_router);
// Configure bandwidth estimation.
- channel_proxy_->RegisterReceiverCongestionControlObjects(packet_router);
+ channel_receive_->RegisterReceiverCongestionControlObjects(packet_router);
// Register with transport.
rtp_stream_receiver_ = receiver_controller->CreateReceiver(
- config.rtp.remote_ssrc, channel_proxy_.get());
+ config.rtp.remote_ssrc, channel_receive_.get());
}
ConfigureStream(this, config, true);
}
@@ -136,9 +134,9 @@
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RTC_LOG(LS_INFO) << "~AudioReceiveStream: " << config_.rtp.remote_ssrc;
Stop();
- channel_proxy_->DisassociateSendChannel();
+ channel_receive_->SetAssociatedSendChannel(nullptr);
if (!config_.media_transport) {
- channel_proxy_->ResetReceiverCongestionControlObjects();
+ channel_receive_->ResetReceiverCongestionControlObjects();
}
}
@@ -153,7 +151,7 @@
if (playing_) {
return;
}
- channel_proxy_->StartPlayout();
+ channel_receive_->StartPlayout();
playing_ = true;
audio_state()->AddReceivingStream(this);
}
@@ -163,7 +161,7 @@
if (!playing_) {
return;
}
- channel_proxy_->StopPlayout();
+ channel_receive_->StopPlayout();
playing_ = false;
audio_state()->RemoveReceivingStream(this);
}
@@ -174,11 +172,11 @@
stats.remote_ssrc = config_.rtp.remote_ssrc;
webrtc::CallReceiveStatistics call_stats =
- channel_proxy_->GetRTCPStatistics();
+ channel_receive_->GetRTCPStatistics();
// TODO(solenberg): Don't return here if we can't get the codec - return the
// stats we *can* get.
webrtc::CodecInst codec_inst = {0};
- if (!channel_proxy_->GetRecCodec(&codec_inst)) {
+ if (!channel_receive_->GetRecCodec(&codec_inst)) {
return stats;
}
@@ -195,13 +193,13 @@
if (codec_inst.plfreq / 1000 > 0) {
stats.jitter_ms = call_stats.jitterSamples / (codec_inst.plfreq / 1000);
}
- stats.delay_estimate_ms = channel_proxy_->GetDelayEstimate();
- stats.audio_level = channel_proxy_->GetSpeechOutputLevelFullRange();
- stats.total_output_energy = channel_proxy_->GetTotalOutputEnergy();
- stats.total_output_duration = channel_proxy_->GetTotalOutputDuration();
+ stats.delay_estimate_ms = channel_receive_->GetDelayEstimate();
+ stats.audio_level = channel_receive_->GetSpeechOutputLevelFullRange();
+ stats.total_output_energy = channel_receive_->GetTotalOutputEnergy();
+ stats.total_output_duration = channel_receive_->GetTotalOutputDuration();
// Get jitter buffer and total delay (alg + jitter + playout) stats.
- auto ns = channel_proxy_->GetNetworkStatistics();
+ auto ns = channel_receive_->GetNetworkStatistics();
stats.jitter_buffer_ms = ns.currentBufferSize;
stats.jitter_buffer_preferred_ms = ns.preferredBufferSize;
stats.total_samples_received = ns.totalSamplesReceived;
@@ -217,7 +215,7 @@
stats.accelerate_rate = Q14ToFloat(ns.currentAccelerateRate);
stats.preemptive_expand_rate = Q14ToFloat(ns.currentPreemptiveRate);
- auto ds = channel_proxy_->GetDecodingCallStatistics();
+ auto ds = channel_receive_->GetDecodingCallStatistics();
stats.decoding_calls_to_silence_generator = ds.calls_to_silence_generator;
stats.decoding_calls_to_neteq = ds.calls_to_neteq;
stats.decoding_normal = ds.decoded_normal;
@@ -231,23 +229,23 @@
void AudioReceiveStream::SetSink(AudioSinkInterface* sink) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
- channel_proxy_->SetSink(sink);
+ channel_receive_->SetSink(sink);
}
void AudioReceiveStream::SetGain(float gain) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
- channel_proxy_->SetChannelOutputVolumeScaling(gain);
+ channel_receive_->SetChannelOutputVolumeScaling(gain);
}
std::vector<RtpSource> AudioReceiveStream::GetSources() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
- return channel_proxy_->GetSources();
+ return channel_receive_->GetSources();
}
AudioMixer::Source::AudioFrameInfo AudioReceiveStream::GetAudioFrameWithInfo(
int sample_rate_hz,
AudioFrame* audio_frame) {
- return channel_proxy_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame);
+ return channel_receive_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame);
}
int AudioReceiveStream::Ssrc() const {
@@ -255,7 +253,7 @@
}
int AudioReceiveStream::PreferredSampleRate() const {
- return channel_proxy_->PreferredSampleRate();
+ return channel_receive_->PreferredSampleRate();
}
int AudioReceiveStream::id() const {
@@ -265,32 +263,29 @@
absl::optional<Syncable::Info> AudioReceiveStream::GetInfo() const {
RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
- absl::optional<Syncable::Info> info = channel_proxy_->GetSyncInfo();
+ absl::optional<Syncable::Info> info = channel_receive_->GetSyncInfo();
if (!info)
return absl::nullopt;
- info->current_delay_ms = channel_proxy_->GetDelayEstimate();
+ info->current_delay_ms = channel_receive_->GetDelayEstimate();
return info;
}
uint32_t AudioReceiveStream::GetPlayoutTimestamp() const {
// Called on video capture thread.
- return channel_proxy_->GetPlayoutTimestamp();
+ return channel_receive_->GetPlayoutTimestamp();
}
void AudioReceiveStream::SetMinimumPlayoutDelay(int delay_ms) {
RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
- return channel_proxy_->SetMinimumPlayoutDelay(delay_ms);
+ return channel_receive_->SetMinimumPlayoutDelay(delay_ms);
}
void AudioReceiveStream::AssociateSendStream(AudioSendStream* send_stream) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
- if (send_stream) {
- channel_proxy_->AssociateSendChannel(send_stream->GetChannelProxy());
- } else {
- channel_proxy_->DisassociateSendChannel();
- }
+ channel_receive_->SetAssociatedSendChannel(
+ send_stream ? send_stream->GetChannel() : nullptr);
associated_send_stream_ = send_stream;
}
@@ -303,7 +298,7 @@
// calls on the worker thread. We should move towards always using a network
// thread. Then this check can be enabled.
// RTC_DCHECK(!thread_checker_.CalledOnValidThread());
- return channel_proxy_->ReceivedRTCPPacket(packet, length);
+ return channel_receive_->ReceivedRTCPPacket(packet, length);
}
void AudioReceiveStream::OnRtpPacket(const RtpPacketReceived& packet) {
@@ -311,7 +306,7 @@
// calls on the worker thread. We should move towards always using a network
// thread. Then this check can be enabled.
// RTC_DCHECK(!thread_checker_.CalledOnValidThread());
- channel_proxy_->OnRtpPacket(packet);
+ channel_receive_->OnRtpPacket(packet);
}
const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
@@ -337,7 +332,7 @@
RTC_LOG(LS_INFO) << "AudioReceiveStream::ConfigureStream: "
<< new_config.ToString();
RTC_DCHECK(stream);
- const auto& channel_proxy = stream->channel_proxy_;
+ const auto& channel_receive = stream->channel_receive_;
const auto& old_config = stream->config_;
// Configuration parameters which cannot be changed.
@@ -351,7 +346,7 @@
old_config.decoder_factory == new_config.decoder_factory);
if (first_time || old_config.rtp.local_ssrc != new_config.rtp.local_ssrc) {
- channel_proxy->SetLocalSSRC(new_config.rtp.local_ssrc);
+ channel_receive->SetLocalSSRC(new_config.rtp.local_ssrc);
}
if (!first_time) {
@@ -363,11 +358,11 @@
// using the actual packet size for the configured codec.
if (first_time || old_config.rtp.nack.rtp_history_ms !=
new_config.rtp.nack.rtp_history_ms) {
- channel_proxy->SetNACKStatus(new_config.rtp.nack.rtp_history_ms != 0,
- new_config.rtp.nack.rtp_history_ms / 20);
+ channel_receive->SetNACKStatus(new_config.rtp.nack.rtp_history_ms != 0,
+ new_config.rtp.nack.rtp_history_ms / 20);
}
if (first_time || old_config.decoder_map != new_config.decoder_map) {
- channel_proxy->SetReceiveCodecs(new_config.decoder_map);
+ channel_receive->SetReceiveCodecs(new_config.decoder_map);
}
stream->config_ = new_config;
