Update to current webrtc library

This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.

webrtc/api/audio_codecs/audio_encoder.cc

@@ -0,0 +1,113 @@
+/*
+ *  Copyright (c) 2014 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.
+ */
+
+#include "api/audio_codecs/audio_encoder.h"
+
+#include "rtc_base/checks.h"
+#include "rtc_base/trace_event.h"
+
+namespace webrtc {
+
+ANAStats::ANAStats() = default;
+ANAStats::~ANAStats() = default;
+ANAStats::ANAStats(const ANAStats&) = default;
+
+AudioEncoder::EncodedInfo::EncodedInfo() = default;
+AudioEncoder::EncodedInfo::EncodedInfo(const EncodedInfo&) = default;
+AudioEncoder::EncodedInfo::EncodedInfo(EncodedInfo&&) = default;
+AudioEncoder::EncodedInfo::~EncodedInfo() = default;
+AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(
+    const EncodedInfo&) = default;
+AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(EncodedInfo&&) =
+    default;
+
+int AudioEncoder::RtpTimestampRateHz() const {
+  return SampleRateHz();
+}
+
+AudioEncoder::EncodedInfo AudioEncoder::Encode(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  TRACE_EVENT0("webrtc", "AudioEncoder::Encode");
+  RTC_CHECK_EQ(audio.size(),
+               static_cast<size_t>(NumChannels() * SampleRateHz() / 100));
+
+  const size_t old_size = encoded->size();
+  EncodedInfo info = EncodeImpl(rtp_timestamp, audio, encoded);
+  RTC_CHECK_EQ(encoded->size() - old_size, info.encoded_bytes);
+  return info;
+}
+
+bool AudioEncoder::SetFec(bool enable) {
+  return !enable;
+}
+
+bool AudioEncoder::SetDtx(bool enable) {
+  return !enable;
+}
+
+bool AudioEncoder::GetDtx() const {
+  return false;
+}
+
+bool AudioEncoder::SetApplication(Application application) {
+  return false;
+}
+
+void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {}
+
+void AudioEncoder::SetTargetBitrate(int target_bps) {}
+
+rtc::ArrayView<std::unique_ptr<AudioEncoder>>
+AudioEncoder::ReclaimContainedEncoders() {
+  return nullptr;
+}
+
+bool AudioEncoder::EnableAudioNetworkAdaptor(const std::string& config_string,
+                                             RtcEventLog* event_log) {
+  return false;
+}
+
+void AudioEncoder::DisableAudioNetworkAdaptor() {}
+
+void AudioEncoder::OnReceivedUplinkPacketLossFraction(
+    float uplink_packet_loss_fraction) {}
+
+void AudioEncoder::OnReceivedUplinkRecoverablePacketLossFraction(
+    float uplink_recoverable_packet_loss_fraction) {
+  RTC_NOTREACHED();
+}
+
+void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) {
+  OnReceivedUplinkBandwidth(target_audio_bitrate_bps, absl::nullopt);
+}
+
+void AudioEncoder::OnReceivedUplinkBandwidth(
+    int target_audio_bitrate_bps,
+    absl::optional<int64_t> bwe_period_ms) {}
+
+void AudioEncoder::OnReceivedUplinkAllocation(BitrateAllocationUpdate update) {
+  OnReceivedUplinkBandwidth(update.target_bitrate.bps(),
+                            update.bwe_period.ms());
+}
+
+void AudioEncoder::OnReceivedRtt(int rtt_ms) {}
+
+void AudioEncoder::OnReceivedOverhead(size_t overhead_bytes_per_packet) {}
+
+void AudioEncoder::SetReceiverFrameLengthRange(int min_frame_length_ms,
+                                               int max_frame_length_ms) {}
+
+ANAStats AudioEncoder::GetANAStats() const {
+  return ANAStats();
+}
+
+}  // namespace webrtc