Adding upstream version 115.7.0esr.upstream/115.7.0esr

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

1 files changed, 210 insertions, 0 deletions

diff --git a/third_party/libwebrtc/call/audio_receive_stream.h b/third_party/libwebrtc/call/audio_receive_stream.h
new file mode 100644
index 0000000000..6fc93b2d9a
--- /dev/null
+++ b/third_party/libwebrtc/call/audio_receive_stream.h
@@ -0,0 +1,210 @@
+/*
+ *  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 CALL_AUDIO_RECEIVE_STREAM_H_
+#define CALL_AUDIO_RECEIVE_STREAM_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/call/transport.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "api/crypto/crypto_options.h"
+#include "api/rtp_parameters.h"
+#include "call/receive_stream.h"
+#include "call/rtp_config.h"
+
+namespace webrtc {
+class AudioSinkInterface;
+
+class AudioReceiveStreamInterface : public MediaReceiveStreamInterface {
+ public:
+  struct Stats {
+    Stats();
+    ~Stats();
+    uint32_t remote_ssrc = 0;
+    int64_t payload_bytes_rcvd = 0;
+    int64_t header_and_padding_bytes_rcvd = 0;
+    uint32_t packets_rcvd = 0;
+    uint64_t fec_packets_received = 0;
+    uint64_t fec_packets_discarded = 0;
+    int32_t packets_lost = 0;
+    uint64_t packets_discarded = 0;
+    uint32_t nacks_sent = 0;
+    std::string codec_name;
+    absl::optional<int> codec_payload_type;
+    uint32_t jitter_ms = 0;
+    uint32_t jitter_buffer_ms = 0;
+    uint32_t jitter_buffer_preferred_ms = 0;
+    uint32_t delay_estimate_ms = 0;
+    int32_t audio_level = -1;
+    // Stats below correspond to similarly-named fields in the WebRTC stats
+    // spec. https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats
+    double total_output_energy = 0.0;
+    uint64_t total_samples_received = 0;
+    double total_output_duration = 0.0;
+    uint64_t concealed_samples = 0;
+    uint64_t silent_concealed_samples = 0;
+    uint64_t concealment_events = 0;
+    double jitter_buffer_delay_seconds = 0.0;
+    uint64_t jitter_buffer_emitted_count = 0;
+    double jitter_buffer_target_delay_seconds = 0.0;
+    double jitter_buffer_minimum_delay_seconds = 0.0;
+    uint64_t inserted_samples_for_deceleration = 0;
+    uint64_t removed_samples_for_acceleration = 0;
+    // Stats below DO NOT correspond directly to anything in the WebRTC stats
+    float expand_rate = 0.0f;
+    float speech_expand_rate = 0.0f;
+    float secondary_decoded_rate = 0.0f;
+    float secondary_discarded_rate = 0.0f;
+    float accelerate_rate = 0.0f;
+    float preemptive_expand_rate = 0.0f;
+    uint64_t delayed_packet_outage_samples = 0;
+    int32_t decoding_calls_to_silence_generator = 0;
+    int32_t decoding_calls_to_neteq = 0;
+    int32_t decoding_normal = 0;
+    // TODO(alexnarest): Consider decoding_neteq_plc for consistency
+    int32_t decoding_plc = 0;
+    int32_t decoding_codec_plc = 0;
+    int32_t decoding_cng = 0;
+    int32_t decoding_plc_cng = 0;
+    int32_t decoding_muted_output = 0;
+    int64_t capture_start_ntp_time_ms = 0;
+    // The timestamp at which the last packet was received, i.e. the time of the
+    // local clock when it was received - not the RTP timestamp of that packet.
+    // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-lastpacketreceivedtimestamp
+    absl::optional<int64_t> last_packet_received_timestamp_ms;
+    uint64_t jitter_buffer_flushes = 0;
+    double relative_packet_arrival_delay_seconds = 0.0;
+    int32_t interruption_count = 0;
+    int32_t total_interruption_duration_ms = 0;
+    // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-estimatedplayouttimestamp
+    absl::optional<int64_t> estimated_playout_ntp_timestamp_ms;
+    // Remote outbound stats derived by the received RTCP sender reports.
+    // https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*
+    absl::optional<int64_t> last_sender_report_timestamp_ms;
+    absl::optional<int64_t> last_sender_report_remote_timestamp_ms;
+    uint32_t sender_reports_packets_sent = 0;
+    uint64_t sender_reports_bytes_sent = 0;
+    uint64_t sender_reports_reports_count = 0;
+    absl::optional<TimeDelta> round_trip_time;
+    TimeDelta total_round_trip_time = TimeDelta::Zero();
+    int round_trip_time_measurements;
+  };
+
+  struct Config {
+    Config();
+    ~Config();
+
+    std::string ToString() const;
+
+    // Receive-stream specific RTP settings.
+    struct Rtp : public ReceiveStreamRtpConfig {
+      Rtp();
+      ~Rtp();
+
+      std::string ToString() const;
+
+      // See NackConfig for description.
+      NackConfig nack;
+
+      RtcpEventObserver* rtcp_event_observer = nullptr;
+    } rtp;
+
+    // Receive-side RTT.
+    bool enable_non_sender_rtt = false;
+
+    Transport* rtcp_send_transport = nullptr;
+
+    // NetEq settings.
+    size_t jitter_buffer_max_packets = 200;
+    bool jitter_buffer_fast_accelerate = false;
+    int jitter_buffer_min_delay_ms = 0;
+
+    // Identifier for an A/V synchronization group. Empty string to disable.
+    // TODO(pbos): Synchronize streams in a sync group, not just one video
+    // stream to one audio stream. Tracked by issue webrtc:4762.
+    std::string sync_group;
+
+    // Decoder specifications for every payload type that we can receive.
+    std::map<int, SdpAudioFormat> decoder_map;
+
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory;
+
+    absl::optional<AudioCodecPairId> codec_pair_id;
+
+    // Per PeerConnection crypto options.
+    webrtc::CryptoOptions crypto_options;
+
+    // An optional custom frame decryptor that allows the entire frame to be
+    // decrypted in whatever way the caller choses. This is not required by
+    // default.
+    // TODO(tommi): Remove this member variable from the struct. It's not
+    // a part of the AudioReceiveStreamInterface state but rather a pass through
+    // variable.
+    rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor;
+
+    // An optional frame transformer used by insertable streams to transform
+    // encoded frames.
+    // TODO(tommi): Remove this member variable from the struct. It's not
+    // a part of the AudioReceiveStreamInterface state but rather a pass through
+    // variable.
+    rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer;
+  };
+
+  // Methods that support reconfiguring the stream post initialization.
+  virtual void SetDecoderMap(std::map<int, SdpAudioFormat> decoder_map) = 0;
+  virtual void SetNackHistory(int history_ms) = 0;
+  virtual void SetNonSenderRttMeasurement(bool enabled) = 0;
+
+  // Returns true if the stream has been started.
+  virtual bool IsRunning() const = 0;
+
+  virtual Stats GetStats(bool get_and_clear_legacy_stats) const = 0;
+  Stats GetStats() { return GetStats(/*get_and_clear_legacy_stats=*/true); }
+
+  // Sets an audio sink that receives unmixed audio from the receive stream.
+  // Ownership of the sink is managed by the caller.
+  // Only one sink can be set and passing a null sink clears an existing one.
+  // NOTE: Audio must still somehow be pulled through AudioTransport for audio
+  // to stream through this sink. In practice, this happens if mixed audio
+  // is being pulled+rendered and/or if audio is being pulled for the purposes
+  // of feeding to the AEC.
+  virtual void SetSink(AudioSinkInterface* sink) = 0;
+
+  // Sets playback gain of the stream, applied when mixing, and thus after it
+  // is potentially forwarded to any attached AudioSinkInterface implementation.
+  virtual void SetGain(float gain) = 0;
+
+  // Sets a base minimum for the playout delay. Base minimum delay sets lower
+  // bound on minimum delay value determining lower bound on playout delay.
+  //
+  // Returns true if value was successfully set, false overwise.
+  virtual bool SetBaseMinimumPlayoutDelayMs(int delay_ms) = 0;
+
+  // Returns current value of base minimum delay in milliseconds.
+  virtual int GetBaseMinimumPlayoutDelayMs() const = 0;
+
+  // Synchronization source (stream identifier) to be received.
+  // This member will not change mid-stream and can be assumed to be const
+  // post initialization.
+  virtual uint32_t remote_ssrc() const = 0;
+
+ protected:
+  virtual ~AudioReceiveStreamInterface() {}
+};
+
+}  // namespace webrtc
+
+#endif  // CALL_AUDIO_RECEIVE_STREAM_H_