Adding upstream version 124.0.1.upstream/124.0.1

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

2 files changed, 283 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/include/audio_coding_module.h b/third_party/libwebrtc/modules/audio_coding/include/audio_coding_module.h
new file mode 100644
index 0000000000..5c5bd167c0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/include/audio_coding_module.h
@@ -0,0 +1,146 @@
+/*
+ *  Copyright (c) 2012 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 MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
+#define MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
+
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/function_view.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+
+namespace webrtc {
+
+// forward declarations
+class AudioDecoder;
+class AudioEncoder;
+class AudioFrame;
+struct RTPHeader;
+
+// Callback class used for sending data ready to be packetized
+class AudioPacketizationCallback {
+ public:
+  virtual ~AudioPacketizationCallback() {}
+
+  virtual int32_t SendData(AudioFrameType frame_type,
+                           uint8_t payload_type,
+                           uint32_t timestamp,
+                           const uint8_t* payload_data,
+                           size_t payload_len_bytes,
+                           int64_t absolute_capture_timestamp_ms) {
+    // TODO(bugs.webrtc.org/10739): Deprecate the old SendData and make this one
+    // pure virtual.
+    return SendData(frame_type, payload_type, timestamp, payload_data,
+                    payload_len_bytes);
+  }
+  virtual int32_t SendData(AudioFrameType frame_type,
+                           uint8_t payload_type,
+                           uint32_t timestamp,
+                           const uint8_t* payload_data,
+                           size_t payload_len_bytes) {
+    RTC_DCHECK_NOTREACHED() << "This method must be overridden, or not used.";
+    return -1;
+  }
+};
+
+class AudioCodingModule {
+ protected:
+  AudioCodingModule() {}
+
+ public:
+  static std::unique_ptr<AudioCodingModule> Create();
+  virtual ~AudioCodingModule() = default;
+
+  // `modifier` is called exactly once with one argument: a pointer to the
+  // unique_ptr that holds the current encoder (which is null if there is no
+  // current encoder). For the duration of the call, `modifier` has exclusive
+  // access to the unique_ptr; it may call the encoder, steal the encoder and
+  // replace it with another encoder or with nullptr, etc.
+  virtual void ModifyEncoder(
+      rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) = 0;
+
+  // Utility method for simply replacing the existing encoder with a new one.
+  void SetEncoder(std::unique_ptr<AudioEncoder> new_encoder) {
+    ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
+      *encoder = std::move(new_encoder);
+    });
+  }
+
+  // int32_t RegisterTransportCallback()
+  // Register a transport callback which will be called to deliver
+  // the encoded buffers whenever Process() is called and a
+  // bit-stream is ready.
+  //
+  // Input:
+  //   -transport          : pointer to the callback class
+  //                         transport->SendData() is called whenever
+  //                         Process() is called and bit-stream is ready
+  //                         to deliver.
+  //
+  // Return value:
+  //   -1 if the transport callback could not be registered
+  //    0 if registration is successful.
+  //
+  virtual int32_t RegisterTransportCallback(
+      AudioPacketizationCallback* transport) = 0;
+
+  ///////////////////////////////////////////////////////////////////////////
+  // int32_t Add10MsData()
+  // Add 10MS of raw (PCM) audio data and encode it. If the sampling
+  // frequency of the audio does not match the sampling frequency of the
+  // current encoder ACM will resample the audio. If an encoded packet was
+  // produced, it will be delivered via the callback object registered using
+  // RegisterTransportCallback, and the return value from this function will
+  // be the number of bytes encoded.
+  //
+  // Input:
+  //   -audio_frame        : the input audio frame, containing raw audio
+  //                         sampling frequency etc.
+  //
+  // Return value:
+  //   >= 0   number of bytes encoded.
+  //     -1   some error occurred.
+  //
+  virtual int32_t Add10MsData(const AudioFrame& audio_frame) = 0;
+
+  ///////////////////////////////////////////////////////////////////////////
+  // int SetPacketLossRate()
+  // Sets expected packet loss rate for encoding. Some encoders provide packet
+  // loss gnostic encoding to make stream less sensitive to packet losses,
+  // through e.g., FEC. No effects on codecs that do not provide such encoding.
+  //
+  // Input:
+  //   -packet_loss_rate   : expected packet loss rate (0 -- 100 inclusive).
+  //
+  // Return value
+  //   -1 if failed to set packet loss rate,
+  //   0 if succeeded.
+  //
+  // This is only used in test code that rely on old ACM APIs.
+  // TODO(minyue): Remove it when possible.
+  virtual int SetPacketLossRate(int packet_loss_rate) = 0;
+
+  ///////////////////////////////////////////////////////////////////////////
+  //   statistics
+  //
+
+  virtual ANAStats GetANAStats() const = 0;
+
+  virtual int GetTargetBitrate() const = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/include/audio_coding_module_typedefs.h b/third_party/libwebrtc/modules/audio_coding/include/audio_coding_module_typedefs.h
new file mode 100644
index 0000000000..4b880fb633
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/include/audio_coding_module_typedefs.h
@@ -0,0 +1,137 @@
+/*
+ *  Copyright (c) 2011 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 MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
+#define MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
+
+#include <map>
+
+namespace webrtc {
+
+///////////////////////////////////////////////////////////////////////////
+// enum ACMVADMode
+// An enumerator for aggressiveness of VAD
+// -VADNormal                : least aggressive mode.
+// -VADLowBitrate            : more aggressive than "VADNormal" to save on
+//                             bit-rate.
+// -VADAggr                  : an aggressive mode.
+// -VADVeryAggr              : the most agressive mode.
+//
+enum ACMVADMode {
+  VADNormal = 0,
+  VADLowBitrate = 1,
+  VADAggr = 2,
+  VADVeryAggr = 3
+};
+
+enum class AudioFrameType {
+  kEmptyFrame = 0,
+  kAudioFrameSpeech = 1,
+  kAudioFrameCN = 2,
+};
+
+///////////////////////////////////////////////////////////////////////////
+//
+// Enumeration of Opus mode for intended application.
+//
+// kVoip              : optimized for voice signals.
+// kAudio             : optimized for non-voice signals like music.
+//
+enum OpusApplicationMode {
+  kVoip = 0,
+  kAudio = 1,
+};
+
+// Statistics for calls to AudioCodingModule::PlayoutData10Ms().
+struct AudioDecodingCallStats {
+  AudioDecodingCallStats()
+      : calls_to_silence_generator(0),
+        calls_to_neteq(0),
+        decoded_normal(0),
+        decoded_neteq_plc(0),
+        decoded_codec_plc(0),
+        decoded_cng(0),
+        decoded_plc_cng(0),
+        decoded_muted_output(0) {}
+
+  int calls_to_silence_generator;  // Number of calls where silence generated,
+                                   // and NetEq was disengaged from decoding.
+  int calls_to_neteq;              // Number of calls to NetEq.
+  int decoded_normal;     // Number of calls where audio RTP packet decoded.
+  int decoded_neteq_plc;  // Number of calls resulted in NetEq PLC.
+  int decoded_codec_plc;  // Number of calls resulted in codec PLC.
+  int decoded_cng;  // Number of calls where comfort noise generated due to DTX.
+  int decoded_plc_cng;       // Number of calls resulted where PLC faded to CNG.
+  int decoded_muted_output;  // Number of calls returning a muted state output.
+};
+
+// NETEQ statistics.
+struct NetworkStatistics {
+  // current jitter buffer size in ms
+  uint16_t currentBufferSize;
+  // preferred (optimal) buffer size in ms
+  uint16_t preferredBufferSize;
+  // adding extra delay due to "peaky jitter"
+  bool jitterPeaksFound;
+  // Stats below correspond to similarly-named fields in the WebRTC stats spec.
+  // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats
+  uint64_t totalSamplesReceived;
+  uint64_t concealedSamples;
+  uint64_t silentConcealedSamples;
+  uint64_t concealmentEvents;
+  uint64_t jitterBufferDelayMs;
+  uint64_t jitterBufferTargetDelayMs;
+  uint64_t jitterBufferMinimumDelayMs;
+  uint64_t jitterBufferEmittedCount;
+  uint64_t insertedSamplesForDeceleration;
+  uint64_t removedSamplesForAcceleration;
+  uint64_t fecPacketsReceived;
+  uint64_t fecPacketsDiscarded;
+  // Stats below correspond to similarly-named fields in the WebRTC stats spec.
+  // https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats
+  uint64_t packetsDiscarded;
+  // Stats below DO NOT correspond directly to anything in the WebRTC stats
+  // fraction (of original stream) of synthesized audio inserted through
+  // expansion (in Q14)
+  uint16_t currentExpandRate;
+  // fraction (of original stream) of synthesized speech inserted through
+  // expansion (in Q14)
+  uint16_t currentSpeechExpandRate;
+  // fraction of synthesized speech inserted through pre-emptive expansion
+  // (in Q14)
+  uint16_t currentPreemptiveRate;
+  // fraction of data removed through acceleration (in Q14)
+  uint16_t currentAccelerateRate;
+  // fraction of data coming from secondary decoding (in Q14)
+  uint16_t currentSecondaryDecodedRate;
+  // Fraction of secondary data, including FEC and RED, that is discarded (in
+  // Q14). Discarding of secondary data can be caused by the reception of the
+  // primary data, obsoleting the secondary data. It can also be caused by early
+  // or late arrival of secondary data.
+  uint16_t currentSecondaryDiscardedRate;
+  // average packet waiting time in the jitter buffer (ms)
+  int meanWaitingTimeMs;
+  // max packet waiting time in the jitter buffer (ms)
+  int maxWaitingTimeMs;
+  // count of the number of buffer flushes
+  uint64_t packetBufferFlushes;
+  // number of samples expanded due to delayed packets
+  uint64_t delayedPacketOutageSamples;
+  // arrival delay of incoming packets
+  uint64_t relativePacketArrivalDelayMs;
+  // number of audio interruptions
+  int32_t interruptionCount;
+  // total duration of audio interruptions
+  int32_t totalInterruptionDurationMs;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_