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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/pc/remote_audio_source.cc | |
parent | Initial commit. (diff) | |
download | thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/pc/remote_audio_source.cc')
-rw-r--r-- | third_party/libwebrtc/pc/remote_audio_source.cc | 184 |
1 files changed, 184 insertions, 0 deletions
diff --git a/third_party/libwebrtc/pc/remote_audio_source.cc b/third_party/libwebrtc/pc/remote_audio_source.cc new file mode 100644 index 0000000000..a516c57617 --- /dev/null +++ b/third_party/libwebrtc/pc/remote_audio_source.cc @@ -0,0 +1,184 @@ +/* + * Copyright 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "pc/remote_audio_source.h" + +#include <stddef.h> + +#include <memory> +#include <string> +#include <utility> + +#include "absl/algorithm/container.h" +#include "api/scoped_refptr.h" +#include "api/sequence_checker.h" +#include "api/task_queue/task_queue_base.h" +#include "rtc_base/checks.h" +#include "rtc_base/logging.h" +#include "rtc_base/strings/string_format.h" +#include "rtc_base/trace_event.h" + +namespace webrtc { + +// This proxy is passed to the underlying media engine to receive audio data as +// they come in. The data will then be passed back up to the RemoteAudioSource +// which will fan it out to all the sinks that have been added to it. +class RemoteAudioSource::AudioDataProxy : public AudioSinkInterface { + public: + explicit AudioDataProxy(RemoteAudioSource* source) : source_(source) { + RTC_DCHECK(source); + } + + AudioDataProxy() = delete; + AudioDataProxy(const AudioDataProxy&) = delete; + AudioDataProxy& operator=(const AudioDataProxy&) = delete; + + ~AudioDataProxy() override { source_->OnAudioChannelGone(); } + + // AudioSinkInterface implementation. + void OnData(const AudioSinkInterface::Data& audio) override { + source_->OnData(audio); + } + + private: + const rtc::scoped_refptr<RemoteAudioSource> source_; +}; + +RemoteAudioSource::RemoteAudioSource( + TaskQueueBase* worker_thread, + OnAudioChannelGoneAction on_audio_channel_gone_action) + : main_thread_(TaskQueueBase::Current()), + worker_thread_(worker_thread), + on_audio_channel_gone_action_(on_audio_channel_gone_action), + state_(MediaSourceInterface::kInitializing) { + RTC_DCHECK(main_thread_); + RTC_DCHECK(worker_thread_); +} + +RemoteAudioSource::~RemoteAudioSource() { + RTC_DCHECK(audio_observers_.empty()); + if (!sinks_.empty()) { + RTC_LOG(LS_WARNING) + << "RemoteAudioSource destroyed while sinks_ is non-empty."; + } +} + +void RemoteAudioSource::Start( + cricket::VoiceMediaReceiveChannelInterface* media_channel, + absl::optional<uint32_t> ssrc) { + RTC_DCHECK_RUN_ON(worker_thread_); + + // Register for callbacks immediately before AddSink so that we always get + // notified when a channel goes out of scope (signaled when "AudioDataProxy" + // is destroyed). + RTC_DCHECK(media_channel); + ssrc ? media_channel->SetRawAudioSink(*ssrc, + std::make_unique<AudioDataProxy>(this)) + : media_channel->SetDefaultRawAudioSink( + std::make_unique<AudioDataProxy>(this)); +} + +void RemoteAudioSource::Stop( + cricket::VoiceMediaReceiveChannelInterface* media_channel, + absl::optional<uint32_t> ssrc) { + RTC_DCHECK_RUN_ON(worker_thread_); + RTC_DCHECK(media_channel); + ssrc ? media_channel->SetRawAudioSink(*ssrc, nullptr) + : media_channel->SetDefaultRawAudioSink(nullptr); +} + +void RemoteAudioSource::SetState(SourceState new_state) { + RTC_DCHECK_RUN_ON(main_thread_); + if (state_ != new_state) { + state_ = new_state; + FireOnChanged(); + } +} + +MediaSourceInterface::SourceState RemoteAudioSource::state() const { + RTC_DCHECK_RUN_ON(main_thread_); + return state_; +} + +bool RemoteAudioSource::remote() const { + RTC_DCHECK_RUN_ON(main_thread_); + return true; +} + +void RemoteAudioSource::SetVolume(double volume) { + RTC_DCHECK_GE(volume, 0); + RTC_DCHECK_LE(volume, 10); + RTC_LOG(LS_INFO) << rtc::StringFormat("RAS::%s({volume=%.2f})", __func__, + volume); + for (auto* observer : audio_observers_) { + observer->OnSetVolume(volume); + } +} + +void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) { + RTC_DCHECK(observer != NULL); + RTC_DCHECK(!absl::c_linear_search(audio_observers_, observer)); + audio_observers_.push_back(observer); +} + +void RemoteAudioSource::UnregisterAudioObserver(AudioObserver* observer) { + RTC_DCHECK(observer != NULL); + audio_observers_.remove(observer); +} + +void RemoteAudioSource::AddSink(AudioTrackSinkInterface* sink) { + RTC_DCHECK_RUN_ON(main_thread_); + RTC_DCHECK(sink); + + MutexLock lock(&sink_lock_); + RTC_DCHECK(!absl::c_linear_search(sinks_, sink)); + sinks_.push_back(sink); +} + +void RemoteAudioSource::RemoveSink(AudioTrackSinkInterface* sink) { + RTC_DCHECK_RUN_ON(main_thread_); + RTC_DCHECK(sink); + + MutexLock lock(&sink_lock_); + sinks_.remove(sink); +} + +void RemoteAudioSource::OnData(const AudioSinkInterface::Data& audio) { + // Called on the externally-owned audio callback thread, via/from webrtc. + TRACE_EVENT0("webrtc", "RemoteAudioSource::OnData"); + MutexLock lock(&sink_lock_); + for (auto* sink : sinks_) { + // When peerconnection acts as an audio source, it should not provide + // absolute capture timestamp. + sink->OnData(audio.data, 16, audio.sample_rate, audio.channels, + audio.samples_per_channel, + /*absolute_capture_timestamp_ms=*/absl::nullopt); + } +} + +void RemoteAudioSource::OnAudioChannelGone() { + if (on_audio_channel_gone_action_ != OnAudioChannelGoneAction::kEnd) { + return; + } + // Called when the audio channel is deleted. It may be the worker thread or + // may be a different task queue. + // This object needs to live long enough for the cleanup logic in the posted + // task to run, so take a reference to it. Sometimes the task may not be + // processed (because the task queue was destroyed shortly after this call), + // but that is fine because the task queue destructor will take care of + // destroying task which will release the reference on RemoteAudioSource. + rtc::scoped_refptr<RemoteAudioSource> thiz(this); + main_thread_->PostTask([thiz = std::move(thiz)] { + thiz->sinks_.clear(); + thiz->SetState(MediaSourceInterface::kEnded); + }); +} + +} // namespace webrtc |