Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream

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

1 files changed, 174 insertions, 0 deletions

diff --git a/third_party/libwebrtc/audio/audio_receive_stream.h b/third_party/libwebrtc/audio/audio_receive_stream.h
new file mode 100644
index 0000000000..51514fb6ce
--- /dev/null
+++ b/third_party/libwebrtc/audio/audio_receive_stream.h
@@ -0,0 +1,174 @@
+/*
+ *  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_RECEIVE_STREAM_H_
+#define AUDIO_AUDIO_RECEIVE_STREAM_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/audio/audio_mixer.h"
+#include "api/neteq/neteq_factory.h"
+#include "api/rtp_headers.h"
+#include "api/sequence_checker.h"
+#include "audio/audio_state.h"
+#include "call/audio_receive_stream.h"
+#include "call/syncable.h"
+#include "modules/rtp_rtcp/source/source_tracker.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+class PacketRouter;
+class RtcEventLog;
+class RtpStreamReceiverControllerInterface;
+class RtpStreamReceiverInterface;
+
+namespace voe {
+class ChannelReceiveInterface;
+}  // namespace voe
+
+namespace internal {
+class AudioSendStream;
+}  // namespace internal
+
+class AudioReceiveStreamImpl final : public webrtc::AudioReceiveStreamInterface,
+                                     public AudioMixer::Source,
+                                     public Syncable {
+ public:
+  AudioReceiveStreamImpl(
+      Clock* clock,
+      PacketRouter* packet_router,
+      NetEqFactory* neteq_factory,
+      const webrtc::AudioReceiveStreamInterface::Config& config,
+      const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+      webrtc::RtcEventLog* event_log);
+  // For unit tests, which need to supply a mock channel receive.
+  AudioReceiveStreamImpl(
+      Clock* clock,
+      PacketRouter* packet_router,
+      const webrtc::AudioReceiveStreamInterface::Config& config,
+      const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+      webrtc::RtcEventLog* event_log,
+      std::unique_ptr<voe::ChannelReceiveInterface> channel_receive);
+
+  AudioReceiveStreamImpl() = delete;
+  AudioReceiveStreamImpl(const AudioReceiveStreamImpl&) = delete;
+  AudioReceiveStreamImpl& operator=(const AudioReceiveStreamImpl&) = delete;
+
+  // Destruction happens on the worker thread. Prior to destruction the caller
+  // must ensure that a registration with the transport has been cleared. See
+  // `RegisterWithTransport` for details.
+  // TODO(tommi): As a further improvement to this, performing the full
+  // destruction on the network thread could be made the default.
+  ~AudioReceiveStreamImpl() override;
+
+  // Called on the network thread to register/unregister with the network
+  // transport.
+  void RegisterWithTransport(
+      RtpStreamReceiverControllerInterface* receiver_controller);
+  // If registration has previously been done (via `RegisterWithTransport`) then
+  // `UnregisterFromTransport` must be called prior to destruction, on the
+  // network thread.
+  void UnregisterFromTransport();
+
+  // webrtc::AudioReceiveStreamInterface implementation.
+  void Start() override;
+  void Stop() override;
+  bool IsRunning() const override;
+  void SetDepacketizerToDecoderFrameTransformer(
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
+      override;
+  void SetDecoderMap(std::map<int, SdpAudioFormat> decoder_map) override;
+  void SetNackHistory(int history_ms) override;
+  void SetNonSenderRttMeasurement(bool enabled) override;
+  void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
+                             frame_decryptor) override;
+  void SetRtpExtensions(std::vector<RtpExtension> extensions) override;
+  RtpHeaderExtensionMap GetRtpExtensionMap() const override;
+
+  webrtc::AudioReceiveStreamInterface::Stats GetStats(
+      bool get_and_clear_legacy_stats) const override;
+  void SetSink(AudioSinkInterface* sink) override;
+  void SetGain(float gain) override;
+  bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override;
+  int GetBaseMinimumPlayoutDelayMs() const override;
+  std::vector<webrtc::RtpSource> GetSources() const override;
+
+  // AudioMixer::Source
+  AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
+                                       AudioFrame* audio_frame) override;
+  int Ssrc() const override;
+  int PreferredSampleRate() const override;
+
+  // Syncable
+  uint32_t id() const override;
+  absl::optional<Syncable::Info> GetInfo() const override;
+  bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
+                              int64_t* time_ms) const override;
+  void SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms,
+                                         int64_t time_ms) override;
+  bool SetMinimumPlayoutDelay(int delay_ms) override;
+
+  void AssociateSendStream(internal::AudioSendStream* send_stream);
+  void DeliverRtcp(const uint8_t* packet, size_t length);
+
+  void SetSyncGroup(absl::string_view sync_group);
+
+  void SetLocalSsrc(uint32_t local_ssrc);
+
+  uint32_t local_ssrc() const;
+
+  uint32_t remote_ssrc() const override {
+    // The remote_ssrc member variable of config_ will never change and can be
+    // considered const.
+    return config_.rtp.remote_ssrc;
+  }
+
+  // Returns a reference to the currently set sync group of the stream.
+  // Must be called on the packet delivery thread.
+  const std::string& sync_group() const;
+
+  const AudioSendStream* GetAssociatedSendStreamForTesting() const;
+
+  // TODO(tommi): Remove this method.
+  void ReconfigureForTesting(
+      const webrtc::AudioReceiveStreamInterface::Config& config);
+
+ private:
+  internal::AudioState* audio_state() const;
+
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker worker_thread_checker_;
+  // TODO(bugs.webrtc.org/11993): This checker conceptually represents
+  // operations that belong to the network thread. The Call class is currently
+  // moving towards handling network packets on the network thread and while
+  // that work is ongoing, this checker may in practice represent the worker
+  // thread, but still serves as a mechanism of grouping together concepts
+  // that belong to the network thread. Once the packets are fully delivered
+  // on the network thread, this comment will be deleted.
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker packet_sequence_checker_;
+  webrtc::AudioReceiveStreamInterface::Config config_;
+  rtc::scoped_refptr<webrtc::AudioState> audio_state_;
+  SourceTracker source_tracker_;
+  const std::unique_ptr<voe::ChannelReceiveInterface> channel_receive_;
+  AudioSendStream* associated_send_stream_
+      RTC_GUARDED_BY(packet_sequence_checker_) = nullptr;
+
+  bool playing_ RTC_GUARDED_BY(worker_thread_checker_) = false;
+
+  std::unique_ptr<RtpStreamReceiverInterface> rtp_stream_receiver_
+      RTC_GUARDED_BY(packet_sequence_checker_);
+};
+}  // namespace webrtc
+
+#endif  // AUDIO_AUDIO_RECEIVE_STREAM_H_