Bump to WebRTC M131 release

Ongoing fixes and improvements, transient suppressor is gone. Also,
dropping isac because it doesn't seem to be useful, and is just build
system deadweight now.

Upstream references:

  Version: 131.0.6778.200
  WebRTC: 79aff54b0fa9238ce3518dd9eaf9610cd6f22e82
  Chromium: 2a19506ad24af755f2a215a4c61f775393e0db42

webrtc/api/audio_codecs/audio_decoder.cc

@@ -10,10 +10,15 @@
 
 #include "api/audio_codecs/audio_decoder.h"
 
+#include <cstddef>
+#include <cstdint>
 #include <memory>
+#include <optional>
 #include <utility>
+#include <vector>
 
 #include "api/array_view.h"
+#include "rtc_base/buffer.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/sanitizer.h"
 #include "rtc_base/trace_event.h"
@@ -32,14 +37,14 @@ class OldStyleEncodedFrame final : public AudioDecoder::EncodedAudioFrame {
     return ret < 0 ? 0 : static_cast<size_t>(ret);
   }
 
-  absl::optional<DecodeResult> Decode(
+  std::optional<DecodeResult> Decode(
       rtc::ArrayView<int16_t> decoded) const override {
     auto speech_type = AudioDecoder::kSpeech;
     const int ret = decoder_->Decode(
         payload_.data(), payload_.size(), decoder_->SampleRateHz(),
         decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
-    return ret < 0 ? absl::nullopt
-                   : absl::optional<DecodeResult>(
+    return ret < 0 ? std::nullopt
+                   : std::optional<DecodeResult>(
                          {static_cast<size_t>(ret), speech_type});
   }
 

webrtc/api/audio_codecs/audio_decoder.h

@@ -15,9 +15,9 @@
 #include <stdint.h>
 
 #include <memory>
+#include <optional>
 #include <vector>
 
-#include "absl/types/optional.h"
 #include "api/array_view.h"
 #include "rtc_base/buffer.h"
 
@@ -57,11 +57,11 @@ class AudioDecoder {
 
     // Decodes this frame of audio and writes the result in `decoded`.
     // `decoded` must be large enough to store as many samples as indicated by a
-    // call to Duration() . On success, returns an absl::optional containing the
+    // call to Duration() . On success, returns an std::optional containing the
     // total number of samples across all channels, as well as whether the
     // decoder produced comfort noise or speech. On failure, returns an empty
-    // absl::optional. Decode may be called at most once per frame object.
-    virtual absl::optional<DecodeResult> Decode(
+    // std::optional. Decode may be called at most once per frame object.
+    virtual std::optional<DecodeResult> Decode(
         rtc::ArrayView<int16_t> decoded) const = 0;
   };
 

webrtc/api/audio_codecs/audio_encoder.cc

@@ -10,6 +10,15 @@
 
 #include "api/audio_codecs/audio_encoder.h"
 
+#include <cstddef>
+#include <cstdint>
+#include <memory>
+#include <optional>
+#include <string>
+
+#include "api/array_view.h"
+#include "api/call/bitrate_allocation.h"
+#include "rtc_base/buffer.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/trace_event.h"
 
@@ -87,12 +96,12 @@ void AudioEncoder::OnReceivedUplinkRecoverablePacketLossFraction(
 }
 
 void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) {
-  OnReceivedUplinkBandwidth(target_audio_bitrate_bps, absl::nullopt);
+  OnReceivedUplinkBandwidth(target_audio_bitrate_bps, std::nullopt);
 }
 
 void AudioEncoder::OnReceivedUplinkBandwidth(
     int target_audio_bitrate_bps,
-    absl::optional<int64_t> bwe_period_ms) {}
+    std::optional<int64_t> bwe_period_ms) {}
 
 void AudioEncoder::OnReceivedUplinkAllocation(BitrateAllocationUpdate update) {
   OnReceivedUplinkBandwidth(update.target_bitrate.bps(),

webrtc/api/audio_codecs/audio_encoder.h

@@ -11,15 +11,19 @@
 #ifndef API_AUDIO_CODECS_AUDIO_ENCODER_H_
 #define API_AUDIO_CODECS_AUDIO_ENCODER_H_
 
+#include <stddef.h>
+#include <stdint.h>
+
 #include <memory>
+#include <optional>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/base/attributes.h"
-#include "absl/types/optional.h"
 #include "api/array_view.h"
 #include "api/call/bitrate_allocation.h"
+#include "api/units/data_rate.h"
 #include "api/units/time_delta.h"
 #include "rtc_base/buffer.h"
 
@@ -35,30 +39,30 @@ struct ANAStats {
   // Number of actions taken by the ANA bitrate controller since the start of
   // the call. If this value is not set, it indicates that the bitrate
   // controller is disabled.
-  absl::optional<uint32_t> bitrate_action_counter;
+  std::optional<uint32_t> bitrate_action_counter;
   // Number of actions taken by the ANA channel controller since the start of
   // the call. If this value is not set, it indicates that the channel
   // controller is disabled.
-  absl::optional<uint32_t> channel_action_counter;
+  std::optional<uint32_t> channel_action_counter;
   // Number of actions taken by the ANA DTX controller since the start of the
   // call. If this value is not set, it indicates that the DTX controller is
   // disabled.
-  absl::optional<uint32_t> dtx_action_counter;
+  std::optional<uint32_t> dtx_action_counter;
   // Number of actions taken by the ANA FEC controller since the start of the
   // call. If this value is not set, it indicates that the FEC controller is
   // disabled.
-  absl::optional<uint32_t> fec_action_counter;
+  std::optional<uint32_t> fec_action_counter;
   // Number of times the ANA frame length controller decided to increase the
   // frame length since the start of the call. If this value is not set, it
   // indicates that the frame length controller is disabled.
-  absl::optional<uint32_t> frame_length_increase_counter;
+  std::optional<uint32_t> frame_length_increase_counter;
   // Number of times the ANA frame length controller decided to decrease the
   // frame length since the start of the call. If this value is not set, it
   // indicates that the frame length controller is disabled.
-  absl::optional<uint32_t> frame_length_decrease_counter;
+  std::optional<uint32_t> frame_length_decrease_counter;
   // The uplink packet loss fractions as set by the ANA FEC controller. If this
   // value is not set, it indicates that the ANA FEC controller is not active.
-  absl::optional<float> uplink_packet_loss_fraction;
+  std::optional<float> uplink_packet_loss_fraction;
 };
 
 // This is the interface class for encoders in AudioCoding module. Each codec
@@ -219,7 +223,7 @@ class AudioEncoder {
   // Provides target audio bitrate and corresponding probing interval of
   // the bandwidth estimator to this encoder to allow it to adapt.
   virtual void OnReceivedUplinkBandwidth(int target_audio_bitrate_bps,
-                                         absl::optional<int64_t> bwe_period_ms);
+                                         std::optional<int64_t> bwe_period_ms);
 
   // Provides target audio bitrate and corresponding probing interval of
   // the bandwidth estimator to this encoder to allow it to adapt.
@@ -240,12 +244,19 @@ class AudioEncoder {
   // Get statistics related to audio network adaptation.
   virtual ANAStats GetANAStats() const;
 
-  // The range of frame lengths that are supported or nullopt if there's no sch
-  // information. This is used to calculated the full bitrate range, including
-  // overhead.
-  virtual absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+  // The range of frame lengths that are supported or nullopt if there's no such
+  // information. This is used together with the bitrate range to calculate the
+  // full bitrate range, including overhead.
+  virtual std::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
       const = 0;
 
+  // The range of payload bitrates that are supported. This is used together
+  // with the frame length range to calculate the full bitrate range, including
+  // overhead.
+  virtual std::optional<std::pair<DataRate, DataRate>> GetBitrateRange() const {
+    return std::nullopt;
+  }
+
   // The maximum number of audio channels supported by WebRTC encoders.
   static constexpr int kMaxNumberOfChannels = 24;