Rename AudioReceiveStream to AudioReceiveStreamInterface
Bug: webrtc:7484
Change-Id: I22eaa7a9e082fc575cf7471d7a2f4f706564d54f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262805
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36965}
diff --git a/video/end_to_end_tests/retransmission_tests.cc b/video/end_to_end_tests/retransmission_tests.cc
index d6d0d41..25fd69c 100644
--- a/video/end_to_end_tests/retransmission_tests.cc
+++ b/video/end_to_end_tests/retransmission_tests.cc
@@ -179,9 +179,9 @@
return SEND_PACKET;
}
- void ModifyAudioConfigs(
- AudioSendStream::Config* send_config,
- std::vector<AudioReceiveStream::Config>* receive_configs) override {
+ void ModifyAudioConfigs(AudioSendStream::Config* send_config,
+ std::vector<AudioReceiveStreamInterface::Config>*
+ receive_configs) override {
(*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
local_ssrc_ = (*receive_configs)[0].rtp.local_ssrc;
remote_ssrc_ = (*receive_configs)[0].rtp.remote_ssrc;
diff --git a/video/end_to_end_tests/transport_feedback_tests.cc b/video/end_to_end_tests/transport_feedback_tests.cc
index 499e5cd..99a87a8 100644
--- a/video/end_to_end_tests/transport_feedback_tests.cc
+++ b/video/end_to_end_tests/transport_feedback_tests.cc
@@ -297,9 +297,9 @@
(*receive_configs)[0].rtp.transport_cc = feedback_enabled_;
}
- void ModifyAudioConfigs(
- AudioSendStream::Config* send_config,
- std::vector<AudioReceiveStream::Config>* receive_configs) override {
+ void ModifyAudioConfigs(AudioSendStream::Config* send_config,
+ std::vector<AudioReceiveStreamInterface::Config>*
+ receive_configs) override {
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(
RtpExtension(RtpExtension::kTransportSequenceNumberUri,
@@ -448,9 +448,9 @@
size_t GetNumVideoStreams() const override { return 1; }
size_t GetNumAudioStreams() const override { return 1; }
- void ModifyAudioConfigs(
- AudioSendStream::Config* send_config,
- std::vector<AudioReceiveStream::Config>* receive_configs) override {
+ void ModifyAudioConfigs(AudioSendStream::Config* send_config,
+ std::vector<AudioReceiveStreamInterface::Config>*
+ receive_configs) override {
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(
RtpExtension(RtpExtension::kTransportSequenceNumberUri,
diff --git a/video/video_analyzer.cc b/video/video_analyzer.cc
index 076ccc3..f298f33 100644
--- a/video/video_analyzer.cc
+++ b/video/video_analyzer.cc
@@ -194,7 +194,8 @@
receive_stream_ = stream;
}
-void VideoAnalyzer::SetAudioReceiveStream(AudioReceiveStream* recv_stream) {
+void VideoAnalyzer::SetAudioReceiveStream(
+ AudioReceiveStreamInterface* recv_stream) {
MutexLock lock(&lock_);
RTC_CHECK(!audio_receive_stream_);
audio_receive_stream_ = recv_stream;
@@ -526,7 +527,7 @@
}
if (audio_receive_stream_ != nullptr) {
- AudioReceiveStream::Stats receive_stats =
+ AudioReceiveStreamInterface::Stats receive_stats =
audio_receive_stream_->GetStats(/*get_and_clear_legacy_stats=*/true);
audio_expand_rate_.AddSample(receive_stats.expand_rate);
audio_accelerate_rate_.AddSample(receive_stats.accelerate_rate);
diff --git a/video/video_analyzer.h b/video/video_analyzer.h
index 3b44f3b..725dacf 100644
--- a/video/video_analyzer.h
+++ b/video/video_analyzer.h
@@ -62,7 +62,7 @@
void SetCall(Call* call);
void SetSendStream(VideoSendStream* stream);
void SetReceiveStream(VideoReceiveStreamInterface* stream);
- void SetAudioReceiveStream(AudioReceiveStream* recv_stream);
+ void SetAudioReceiveStream(AudioReceiveStreamInterface* recv_stream);
rtc::VideoSinkInterface<VideoFrame>* InputInterface();
rtc::VideoSourceInterface<VideoFrame>* OutputInterface();
@@ -222,7 +222,7 @@
Call* call_;
VideoSendStream* send_stream_;
VideoReceiveStreamInterface* receive_stream_;
- AudioReceiveStream* audio_receive_stream_;
+ AudioReceiveStreamInterface* audio_receive_stream_;
CapturedFrameForwarder captured_frame_forwarder_;
const std::string test_label_;
FILE* const graph_data_output_file_;
diff --git a/video/video_quality_test.cc b/video/video_quality_test.cc
index 956fcae..8c39065 100644
--- a/video/video_quality_test.cc
+++ b/video/video_quality_test.cc
@@ -1145,7 +1145,7 @@
void VideoQualityTest::StartAudioStreams() {
audio_send_stream_->Start();
- for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
+ for (AudioReceiveStreamInterface* audio_recv_stream : audio_receive_streams_)
audio_recv_stream->Start();
}
diff --git a/video/video_send_stream_tests.cc b/video/video_send_stream_tests.cc
index 1c53835..18b448a 100644
--- a/video/video_send_stream_tests.cc
+++ b/video/video_send_stream_tests.cc
@@ -1605,9 +1605,9 @@
(*receive_configs)[0].rtp.transport_cc = true;
}
- void ModifyAudioConfigs(
- AudioSendStream::Config* send_config,
- std::vector<AudioReceiveStream::Config>* receive_configs) override {
+ void ModifyAudioConfigs(AudioSendStream::Config* send_config,
+ std::vector<AudioReceiveStreamInterface::Config>*
+ receive_configs) override {
RTC_DCHECK_RUN_ON(&task_queue_thread_);
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(RtpExtension(
