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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
commit36d22d82aa202bb199967e9512281e9a53db42c9 (patch)
tree105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/pc/remote_audio_source.cc
parentInitial commit. (diff)
downloadfirefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz
firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip
Adding upstream version 115.7.0esr.upstream/115.7.0esr
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.cc184
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
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+++ b/third_party/libwebrtc/pc/remote_audio_source.cc
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+/*
+ * 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