Delete support for enabling adaptive isac mode

This appears unused. If deleted, other code related to isac bandwidth
estimation becomes unused and may be deleted in followup cls.

Bug: webrtc:10098
Change-Id: Ifeac2e90de895b12c337ea28cc33704350b9abf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153667
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29252}

modules/audio_coding/BUILD.gn

@@ -370,15 +370,13 @@ rtc_source_set("ilbc_c") {
   ]
 }
 
-rtc_static_library("isac_common") {
+rtc_source_set("isac_common") {
   poisonous = [ "audio_codecs" ]
   sources = [
     "codecs/isac/audio_decoder_isac_t.h",
     "codecs/isac/audio_decoder_isac_t_impl.h",
     "codecs/isac/audio_encoder_isac_t.h",
     "codecs/isac/audio_encoder_isac_t_impl.h",
-    "codecs/isac/locked_bandwidth_info.cc",
-    "codecs/isac/locked_bandwidth_info.h",
   ]
   deps = [
     ":isac_bwinfo",

modules/audio_coding/codecs/isac/audio_decoder_isac_t.h

@@ -16,7 +16,6 @@
 #include "absl/types/optional.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "api/scoped_refptr.h"
-#include "modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
 #include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
@@ -26,7 +25,6 @@ class AudioDecoderIsacT final : public AudioDecoder {
  public:
   struct Config {
     bool IsOk() const;
-    rtc::scoped_refptr<LockedIsacBandwidthInfo> bwinfo;
     int sample_rate_hz = 16000;
   };
   explicit AudioDecoderIsacT(const Config& config);
@@ -52,7 +50,6 @@ class AudioDecoderIsacT final : public AudioDecoder {
  private:
   typename T::instance_type* isac_state_;
   int sample_rate_hz_;
-  rtc::scoped_refptr<LockedIsacBandwidthInfo> bwinfo_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacT);
 };

modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h

@@ -22,16 +22,11 @@ bool AudioDecoderIsacT<T>::Config::IsOk() const {
 
 template <typename T>
 AudioDecoderIsacT<T>::AudioDecoderIsacT(const Config& config)
-    : sample_rate_hz_(config.sample_rate_hz), bwinfo_(config.bwinfo) {
+    : sample_rate_hz_(config.sample_rate_hz) {
   RTC_CHECK(config.IsOk()) << "Unsupported sample rate "
                            << config.sample_rate_hz;
   RTC_CHECK_EQ(0, T::Create(&isac_state_));
   T::DecoderInit(isac_state_);
-  if (bwinfo_) {
-    IsacBandwidthInfo bi;
-    T::GetBandwidthInfo(isac_state_, &bi);
-    bwinfo_->Set(bi);
-  }
   RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, sample_rate_hz_));
 }
 
@@ -78,11 +73,6 @@ int AudioDecoderIsacT<T>::IncomingPacket(const uint8_t* payload,
   int ret = T::UpdateBwEstimate(isac_state_, payload, payload_len,
                                 rtp_sequence_number, rtp_timestamp,
                                 arrival_timestamp);
-  if (bwinfo_) {
-    IsacBandwidthInfo bwinfo;
-    T::GetBandwidthInfo(isac_state_, &bwinfo);
-    bwinfo_->Set(bwinfo);
-  }
   return ret;
 }
 

modules/audio_coding/codecs/isac/audio_encoder_isac_t.h

@@ -15,7 +15,6 @@
 
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/scoped_refptr.h"
-#include "modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
 #include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
@@ -30,8 +29,6 @@ class AudioEncoderIsacT final : public AudioEncoder {
   struct Config {
     bool IsOk() const;
 
-    rtc::scoped_refptr<LockedIsacBandwidthInfo> bwinfo;
-
     int payload_type = 103;
     int sample_rate_hz = 16000;
     int frame_size_ms = 30;
@@ -39,14 +36,6 @@ class AudioEncoderIsacT final : public AudioEncoder {
                                      // rate, in bits/s.
     int max_payload_size_bytes = -1;
     int max_bit_rate = -1;
-
-    // If true, the encoder will dynamically adjust frame size and bit rate;
-    // the configured values are then merely the starting point.
-    bool adaptive_mode = false;
-
-    // In adaptive mode, prevent adaptive changes to the frame size. (Not used
-    // in nonadaptive mode.)
-    bool enforce_frame_size = false;
   };
 
   explicit AudioEncoderIsacT(const Config& config);
@@ -74,7 +63,6 @@ class AudioEncoderIsacT final : public AudioEncoder {
 
   Config config_;
   typename T::instance_type* isac_state_ = nullptr;
-  rtc::scoped_refptr<LockedIsacBandwidthInfo> bwinfo_;
 
   // Have we accepted input but not yet emitted it in a packet?
   bool packet_in_progress_ = false;

modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h

@@ -21,8 +21,7 @@ bool AudioEncoderIsacT<T>::Config::IsOk() const {
     return false;
   if (max_payload_size_bytes < 120 && max_payload_size_bytes != -1)
     return false;
-  if (adaptive_mode && !bwinfo)
+
-    return false;
   switch (sample_rate_hz) {
     case 16000:
       if (max_bit_rate > 53400)
@@ -78,8 +77,6 @@ size_t AudioEncoderIsacT<T>::Max10MsFramesInAPacket() const {
 
 template <typename T>
 int AudioEncoderIsacT<T>::GetTargetBitrate() const {
-  if (config_.adaptive_mode)
-    return -1;
   return config_.bit_rate == 0 ? kDefaultBitRate : config_.bit_rate;
 }
 
@@ -93,11 +90,6 @@ AudioEncoder::EncodedInfo AudioEncoderIsacT<T>::EncodeImpl(
     packet_in_progress_ = true;
     packet_timestamp_ = rtp_timestamp;
   }
-  if (bwinfo_) {
-    IsacBandwidthInfo bwinfo = bwinfo_->Get();
-    T::SetBandwidthInfo(isac_state_, &bwinfo);
-  }
-
   size_t encoded_bytes = encoded->AppendData(
       kSufficientEncodeBufferSizeBytes, [&](rtc::ArrayView<uint8_t> encoded) {
         int r = T::Encode(isac_state_, audio.data(), encoded.data());
@@ -131,19 +123,14 @@ template <typename T>
 void AudioEncoderIsacT<T>::RecreateEncoderInstance(const Config& config) {
   RTC_CHECK(config.IsOk());
   packet_in_progress_ = false;
-  bwinfo_ = config.bwinfo;
   if (isac_state_)
     RTC_CHECK_EQ(0, T::Free(isac_state_));
   RTC_CHECK_EQ(0, T::Create(&isac_state_));
-  RTC_CHECK_EQ(0, T::EncoderInit(isac_state_, config.adaptive_mode ? 0 : 1));
+  RTC_CHECK_EQ(0, T::EncoderInit(isac_state_, 1));
   RTC_CHECK_EQ(0, T::SetEncSampRate(isac_state_, config.sample_rate_hz));
   const int bit_rate = config.bit_rate == 0 ? kDefaultBitRate : config.bit_rate;
-  if (config.adaptive_mode) {
-    RTC_CHECK_EQ(0, T::ControlBwe(isac_state_, bit_rate, config.frame_size_ms,
-                                  config.enforce_frame_size));
-  } else {
   RTC_CHECK_EQ(0, T::Control(isac_state_, bit_rate, config.frame_size_ms));
-  }
+
   if (config.max_payload_size_bytes != -1)
     RTC_CHECK_EQ(
         0, T::SetMaxPayloadSize(isac_state_, config.max_payload_size_bytes));

modules/audio_coding/codecs/isac/locked_bandwidth_info.cc

@@ -1,21 +0,0 @@
-/*
- *  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.
- */
-
-#include "modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
-
-namespace webrtc {
-
-LockedIsacBandwidthInfo::LockedIsacBandwidthInfo() : ref_count_(0) {
-  bwinfo_.in_use = 0;
-}
-
-LockedIsacBandwidthInfo::~LockedIsacBandwidthInfo() = default;
-
-}  // namespace webrtc

modules/audio_coding/codecs/isac/locked_bandwidth_info.h

@@ -1,56 +0,0 @@
-/*
- *  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 MODULES_AUDIO_CODING_CODECS_ISAC_LOCKED_BANDWIDTH_INFO_H_
-#define MODULES_AUDIO_CODING_CODECS_ISAC_LOCKED_BANDWIDTH_INFO_H_
-
-#include "modules/audio_coding/codecs/isac/bandwidth_info.h"
-#include "rtc_base/atomic_ops.h"
-#include "rtc_base/critical_section.h"
-#include "rtc_base/thread_annotations.h"
-
-namespace webrtc {
-
-// An IsacBandwidthInfo that's safe to access from multiple threads because
-// it's protected by a mutex.
-class LockedIsacBandwidthInfo final {
- public:
-  LockedIsacBandwidthInfo();
-  ~LockedIsacBandwidthInfo();
-
-  IsacBandwidthInfo Get() const {
-    rtc::CritScope lock(&lock_);
-    return bwinfo_;
-  }
-
-  void Set(const IsacBandwidthInfo& bwinfo) {
-    rtc::CritScope lock(&lock_);
-    bwinfo_ = bwinfo;
-  }
-
-  int AddRef() const { return rtc::AtomicOps::Increment(&ref_count_); }
-
-  int Release() const {
-    const int count = rtc::AtomicOps::Decrement(&ref_count_);
-    if (count == 0) {
-      delete this;
-    }
-    return count;
-  }
-
- private:
-  mutable volatile int ref_count_;
-  rtc::CriticalSection lock_;
-  IsacBandwidthInfo bwinfo_ RTC_GUARDED_BY(lock_);
-};
-
-}  // namespace webrtc
-
-#endif  // MODULES_AUDIO_CODING_CODECS_ISAC_LOCKED_BANDWIDTH_INFO_H_

modules/audio_coding/neteq/audio_decoder_unittest.cc

@@ -353,7 +353,6 @@ class AudioDecoderIsacFloatTest : public AudioDecoderTest {
     AudioEncoderIsacFloatImpl::Config config;
     config.payload_type = payload_type_;
     config.sample_rate_hz = codec_input_rate_hz_;
-    config.adaptive_mode = false;
     config.frame_size_ms =
         1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
     audio_encoder_.reset(new AudioEncoderIsacFloatImpl(config));
@@ -373,7 +372,6 @@ class AudioDecoderIsacSwbTest : public AudioDecoderTest {
     AudioEncoderIsacFloatImpl::Config config;
     config.payload_type = payload_type_;
     config.sample_rate_hz = codec_input_rate_hz_;
-    config.adaptive_mode = false;
     config.frame_size_ms =
         1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
     audio_encoder_.reset(new AudioEncoderIsacFloatImpl(config));
@@ -393,7 +391,6 @@ class AudioDecoderIsacFixTest : public AudioDecoderTest {
     AudioEncoderIsacFixImpl::Config config;
     config.payload_type = payload_type_;
     config.sample_rate_hz = codec_input_rate_hz_;
-    config.adaptive_mode = false;
     config.frame_size_ms =
         1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
     audio_encoder_.reset(new AudioEncoderIsacFixImpl(config));