Adding upstream version 115.7.0esr.upstream/115.7.0esr

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 92 insertions, 0 deletions

diff --git a/third_party/libwebrtc/audio/audio_state.h b/third_party/libwebrtc/audio/audio_state.h
new file mode 100644
index 0000000000..6c2b7aa453
--- /dev/null
+++ b/third_party/libwebrtc/audio/audio_state.h
@@ -0,0 +1,92 @@
+/*
+ *  Copyright (c) 2015 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.
+ */
+
+#ifndef AUDIO_AUDIO_STATE_H_
+#define AUDIO_AUDIO_STATE_H_
+
+#include <map>
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "audio/audio_transport_impl.h"
+#include "call/audio_state.h"
+#include "rtc_base/containers/flat_set.h"
+#include "rtc_base/ref_count.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class AudioSendStream;
+class AudioReceiveStreamInterface;
+
+namespace internal {
+
+class AudioState : public webrtc::AudioState {
+ public:
+  explicit AudioState(const AudioState::Config& config);
+
+  AudioState() = delete;
+  AudioState(const AudioState&) = delete;
+  AudioState& operator=(const AudioState&) = delete;
+
+  ~AudioState() override;
+
+  AudioProcessing* audio_processing() override;
+  AudioTransport* audio_transport() override;
+
+  void SetPlayout(bool enabled) override;
+  void SetRecording(bool enabled) override;
+
+  void SetStereoChannelSwapping(bool enable) override;
+
+  AudioDeviceModule* audio_device_module() {
+    RTC_DCHECK(config_.audio_device_module);
+    return config_.audio_device_module.get();
+  }
+
+  void AddReceivingStream(webrtc::AudioReceiveStreamInterface* stream);
+  void RemoveReceivingStream(webrtc::AudioReceiveStreamInterface* stream);
+
+  void AddSendingStream(webrtc::AudioSendStream* stream,
+                        int sample_rate_hz,
+                        size_t num_channels);
+  void RemoveSendingStream(webrtc::AudioSendStream* stream);
+
+ private:
+  void UpdateAudioTransportWithSendingStreams();
+  void UpdateNullAudioPollerState() RTC_RUN_ON(&thread_checker_);
+
+  SequenceChecker thread_checker_;
+  SequenceChecker process_thread_checker_;
+  const webrtc::AudioState::Config config_;
+  bool recording_enabled_ = true;
+  bool playout_enabled_ = true;
+
+  // Transports mixed audio from the mixer to the audio device and
+  // recorded audio to the sending streams.
+  AudioTransportImpl audio_transport_;
+
+  // Null audio poller is used to continue polling the audio streams if audio
+  // playout is disabled so that audio processing still happens and the audio
+  // stats are still updated.
+  RepeatingTaskHandle null_audio_poller_ RTC_GUARDED_BY(&thread_checker_);
+
+  webrtc::flat_set<webrtc::AudioReceiveStreamInterface*> receiving_streams_;
+  struct StreamProperties {
+    int sample_rate_hz = 0;
+    size_t num_channels = 0;
+  };
+  std::map<webrtc::AudioSendStream*, StreamProperties> sending_streams_;
+};
+}  // namespace internal
+}  // namespace webrtc
+
+#endif  // AUDIO_AUDIO_STATE_H_