Remove CodecInst pt.3

Finally remove CodecInst from common_types.h, including remaining code referencing it.

TBR=kwiberg

Bug: webrtc:7626
Change-Id: I5e6b949ae9093641e33972af8438d1126fc48556
Reviewed-on: https://webrtc-review.googlesource.com/c/114546
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26036}

common_types.h

@@ -14,12 +14,10 @@
 #include <stddef.h>  // For size_t
 #include <cstdint>
 
-#include "absl/strings/match.h"
 // TODO(sprang): Remove this include when all usage includes it directly.
 #include "api/video/video_bitrate_allocation.h"
 // TODO(bugs.webrtc.org/7660): Delete include once downstream code is updated.
 #include "api/video/video_codec_type.h"
-#include "rtc_base/checks.h"
 
 #if defined(_MSC_VER)
 // Disable "new behavior: elements of array will be default initialized"
@@ -158,29 +156,6 @@ class OverheadObserver {
   virtual void OnOverheadChanged(size_t overhead_bytes_per_packet) = 0;
 };
 
-// ==================================================================
-// Voice specific types
-// ==================================================================
-
-// Each codec supported can be described by this structure.
-struct CodecInst {
-  int pltype;
-  char plname[RTP_PAYLOAD_NAME_SIZE];
-  int plfreq;
-  int pacsize;
-  size_t channels;
-  int rate;  // bits/sec unlike {start,min,max}Bitrate elsewhere in this file!
-
-  bool operator==(const CodecInst& other) const {
-    return pltype == other.pltype &&
-           absl::EqualsIgnoreCase(plname, other.plname) &&
-           plfreq == other.plfreq && pacsize == other.pacsize &&
-           channels == other.channels && rate == other.rate;
-  }
-
-  bool operator!=(const CodecInst& other) const { return !(*this == other); }
-};
-
 // RTP
 enum { kRtpCsrcSize = 15 };  // RFC 3550 page 13
 

media/base/h264_profile_level_id.cc

@@ -15,6 +15,7 @@
 #include <cstring>
 
 #include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
 
 namespace webrtc {
 namespace H264 {

modules/audio_coding/BUILD.gn

@@ -14,24 +14,6 @@ if (!build_with_mozilla) {
 
 visibility = [ ":*" ]
 
-rtc_static_library("audio_format_conversion") {
-  visibility += webrtc_default_visibility
-  sources = [
-    "codecs/audio_format_conversion.cc",
-    "codecs/audio_format_conversion.h",
-  ]
-  deps = [
-    "../..:webrtc_common",
-    "../../api:array_view",
-    "../../api/audio_codecs:audio_codecs_api",
-    "../../rtc_base:checks",
-    "../../rtc_base:rtc_base_approved",
-    "../../rtc_base:sanitizer",
-    "//third_party/abseil-cpp/absl/strings",
-    "//third_party/abseil-cpp/absl/types:optional",
-  ]
-}
-
 rtc_source_set("audio_coding_module_typedefs") {
   sources = [
     "include/audio_coding_module_typedefs.h",
@@ -1319,7 +1301,6 @@ if (rtc_include_tests) {
       ":audio_coding",
       ":audio_coding_module_typedefs",
       ":audio_encoder_cng",
-      ":audio_format_conversion",
       ":pcm16b_c",
       ":red",
       ":webrtc_opus_c",
@@ -1398,7 +1379,6 @@ if (rtc_include_tests) {
     deps = audio_coding_deps + [
              "..:module_api",
              ":audio_coding",
-             ":audio_format_conversion",
              "../../api/audio_codecs:audio_codecs_api",
              "../../api/audio_codecs:builtin_audio_decoder_factory",
              ":neteq_tools",
@@ -1422,7 +1402,6 @@ if (rtc_include_tests) {
              "../../api/audio:audio_frame_api",
              "../../rtc_base:checks",
              ":audio_coding",
-             ":audio_format_conversion",
              ":neteq_tools",
              "../../api/audio_codecs:builtin_audio_decoder_factory",
              "../../api/audio_codecs:builtin_audio_encoder_factory",
@@ -2041,7 +2020,6 @@ if (rtc_include_tests) {
       ":audio_coding",
       ":audio_coding_module_typedefs",
       ":audio_encoder_cng",
-      ":audio_format_conversion",
       ":audio_network_adaptor",
       ":g711",
       ":ilbc",

modules/audio_coding/codecs/audio_format_conversion.cc

@@ -1,89 +0,0 @@
-/*
- *  Copyright (c) 2016 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/audio_format_conversion.h"
-
-#include <string.h>
-#include <string>
-#include <utility>
-
-#include "absl/strings/match.h"
-#include "api/array_view.h"
-#include "rtc_base/checks.h"
-#include "rtc_base/sanitizer.h"
-
-namespace webrtc {
-
-namespace {
-
-CodecInst MakeCodecInst(int payload_type,
-                        const char* name,
-                        int sample_rate,
-                        size_t num_channels) {
-  // Create a CodecInst with some fields set. The remaining fields are zeroed,
-  // but we tell MSan to consider them uninitialized.
-  CodecInst ci = {0};
-  rtc::MsanMarkUninitialized(rtc::MakeArrayView(&ci, 1));
-  ci.pltype = payload_type;
-  strncpy(ci.plname, name, sizeof(ci.plname));
-  ci.plname[sizeof(ci.plname) - 1] = '\0';
-  ci.plfreq = sample_rate;
-  ci.channels = num_channels;
-  return ci;
-}
-
-}  // namespace
-
-SdpAudioFormat CodecInstToSdp(const CodecInst& ci) {
-  if (absl::EqualsIgnoreCase(ci.plname, "g722")) {
-    RTC_CHECK_EQ(16000, ci.plfreq);
-    RTC_CHECK(ci.channels == 1 || ci.channels == 2);
-    return {"g722", 8000, ci.channels};
-  } else if (absl::EqualsIgnoreCase(ci.plname, "opus")) {
-    RTC_CHECK_EQ(48000, ci.plfreq);
-    RTC_CHECK(ci.channels == 1 || ci.channels == 2);
-    return ci.channels == 1
-               ? SdpAudioFormat("opus", 48000, 2)
-               : SdpAudioFormat("opus", 48000, 2, {{"stereo", "1"}});
-  } else {
-    return {ci.plname, ci.plfreq, ci.channels};
-  }
-}
-
-CodecInst SdpToCodecInst(int payload_type, const SdpAudioFormat& audio_format) {
-  if (absl::EqualsIgnoreCase(audio_format.name, "g722")) {
-    RTC_CHECK_EQ(8000, audio_format.clockrate_hz);
-    RTC_CHECK(audio_format.num_channels == 1 || audio_format.num_channels == 2);
-    return MakeCodecInst(payload_type, "g722", 16000,
-                         audio_format.num_channels);
-  } else if (absl::EqualsIgnoreCase(audio_format.name, "opus")) {
-    RTC_CHECK_EQ(48000, audio_format.clockrate_hz);
-    RTC_CHECK_EQ(2, audio_format.num_channels);
-    const int num_channels = [&] {
-      auto stereo = audio_format.parameters.find("stereo");
-      if (stereo != audio_format.parameters.end()) {
-        if (stereo->second == "0") {
-          return 1;
-        } else if (stereo->second == "1") {
-          return 2;
-        } else {
-          RTC_CHECK(false);  // Bad stereo parameter.
-        }
-      }
-      return 1;  // Default to mono.
-    }();
-    return MakeCodecInst(payload_type, "opus", 48000, num_channels);
-  } else {
-    return MakeCodecInst(payload_type, audio_format.name.c_str(),
-                         audio_format.clockrate_hz, audio_format.num_channels);
-  }
-}
-
-}  // namespace webrtc

modules/audio_coding/codecs/audio_format_conversion.h

@@ -1,24 +0,0 @@
-/*
- *  Copyright (c) 2016 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_AUDIO_FORMAT_CONVERSION_H_
-#define MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_CONVERSION_H_
-
-#include "api/audio_codecs/audio_format.h"
-#include "common_types.h"  // NOLINT(build/include)
-
-namespace webrtc {
-
-SdpAudioFormat CodecInstToSdp(const CodecInst& codec_inst);
-CodecInst SdpToCodecInst(int payload_type, const SdpAudioFormat& audio_format);
-
-}  // namespace webrtc
-
-#endif  // MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_CONVERSION_H_

modules/audio_coding/codecs/g711/audio_encoder_pcm.cc

@@ -12,25 +12,11 @@
 
 #include <cstdint>
 
-#include "common_types.h"
 #include "modules/audio_coding/codecs/g711/g711_interface.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
 
-namespace {
-
-template <typename T>
-typename T::Config CreateConfig(const CodecInst& codec_inst) {
-  typename T::Config config;
-  config.frame_size_ms = codec_inst.pacsize / 8;
-  config.num_channels = codec_inst.channels;
-  config.payload_type = codec_inst.pltype;
-  return config;
-}
-
-}  // namespace
-
 bool AudioEncoderPcm::Config::IsOk() const {
   return (frame_size_ms % 10 == 0) && (num_channels >= 1);
 }
@@ -103,9 +89,6 @@ void AudioEncoderPcm::Reset() {
   speech_buffer_.clear();
 }
 
-AudioEncoderPcmA::AudioEncoderPcmA(const CodecInst& codec_inst)
-    : AudioEncoderPcmA(CreateConfig<AudioEncoderPcmA>(codec_inst)) {}
-
 size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
                                     size_t input_len,
                                     uint8_t* encoded) {
@@ -120,9 +103,6 @@ AudioEncoder::CodecType AudioEncoderPcmA::GetCodecType() const {
   return AudioEncoder::CodecType::kPcmA;
 }
 
-AudioEncoderPcmU::AudioEncoderPcmU(const CodecInst& codec_inst)
-    : AudioEncoderPcmU(CreateConfig<AudioEncoderPcmU>(codec_inst)) {}
-
 size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
                                     size_t input_len,
                                     uint8_t* encoded) {

modules/audio_coding/codecs/g711/audio_encoder_pcm.h

@@ -69,8 +69,6 @@ class AudioEncoderPcm : public AudioEncoder {
   uint32_t first_timestamp_in_buffer_;
 };
 
-struct CodecInst;
-
 class AudioEncoderPcmA final : public AudioEncoderPcm {
  public:
   struct Config : public AudioEncoderPcm::Config {
@@ -79,7 +77,6 @@ class AudioEncoderPcmA final : public AudioEncoderPcm {
 
   explicit AudioEncoderPcmA(const Config& config)
       : AudioEncoderPcm(config, kSampleRateHz) {}
-  explicit AudioEncoderPcmA(const CodecInst& codec_inst);
 
  protected:
   size_t EncodeCall(const int16_t* audio,
@@ -103,7 +100,6 @@ class AudioEncoderPcmU final : public AudioEncoderPcm {
 
   explicit AudioEncoderPcmU(const Config& config)
       : AudioEncoderPcm(config, kSampleRateHz) {}
-  explicit AudioEncoderPcmU(const CodecInst& codec_inst);
 
  protected:
   size_t EncodeCall(const int16_t* audio,

modules/audio_coding/codecs/g722/audio_encoder_g722.cc

@@ -12,7 +12,6 @@
 
 #include <cstdint>
 
-#include "common_types.h"
 #include "modules/audio_coding/codecs/g722/g722_interface.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_conversions.h"
@@ -23,13 +22,6 @@ namespace {
 
 const size_t kSampleRateHz = 16000;
 
-AudioEncoderG722Config CreateConfig(const CodecInst& codec_inst) {
-  AudioEncoderG722Config config;
-  config.num_channels = rtc::dchecked_cast<int>(codec_inst.channels);
-  config.frame_size_ms = codec_inst.pacsize / 16;
-  return config;
-}
-
 }  // namespace
 
 AudioEncoderG722Impl::AudioEncoderG722Impl(const AudioEncoderG722Config& config,
@@ -52,9 +44,6 @@ AudioEncoderG722Impl::AudioEncoderG722Impl(const AudioEncoderG722Config& config,
   Reset();
 }
 
-AudioEncoderG722Impl::AudioEncoderG722Impl(const CodecInst& codec_inst)
-    : AudioEncoderG722Impl(CreateConfig(codec_inst), codec_inst.pltype) {}
-
 AudioEncoderG722Impl::~AudioEncoderG722Impl() = default;
 
 int AudioEncoderG722Impl::SampleRateHz() const {

modules/audio_coding/codecs/g722/audio_encoder_g722.h

@@ -21,12 +21,9 @@
 
 namespace webrtc {
 
-struct CodecInst;
-
 class AudioEncoderG722Impl final : public AudioEncoder {
  public:
   AudioEncoderG722Impl(const AudioEncoderG722Config& config, int payload_type);
-  explicit AudioEncoderG722Impl(const CodecInst& codec_inst);
   ~AudioEncoderG722Impl() override;
 
   int SampleRateHz() const override;

modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc

@@ -13,7 +13,6 @@
 #include <algorithm>
 #include <cstdint>
 
-#include "common_types.h"
 #include "modules/audio_coding/codecs/ilbc/ilbc.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_conversions.h"
@@ -24,12 +23,6 @@ namespace {
 
 const int kSampleRateHz = 8000;
 
-AudioEncoderIlbcConfig CreateConfig(const CodecInst& codec_inst) {
-  AudioEncoderIlbcConfig config;
-  config.frame_size_ms = codec_inst.pacsize / 8;
-  return config;
-}
-
 int GetIlbcBitrate(int ptime) {
   switch (ptime) {
     case 20:
@@ -58,9 +51,6 @@ AudioEncoderIlbcImpl::AudioEncoderIlbcImpl(const AudioEncoderIlbcConfig& config,
   Reset();
 }
 
-AudioEncoderIlbcImpl::AudioEncoderIlbcImpl(const CodecInst& codec_inst)
-    : AudioEncoderIlbcImpl(CreateConfig(codec_inst), codec_inst.pltype) {}
-
 AudioEncoderIlbcImpl::~AudioEncoderIlbcImpl() {
   RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
 }

modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h

@@ -18,12 +18,9 @@
 
 namespace webrtc {
 
-struct CodecInst;
-
 class AudioEncoderIlbcImpl final : public AudioEncoder {
  public:
   AudioEncoderIlbcImpl(const AudioEncoderIlbcConfig& config, int payload_type);
-  explicit AudioEncoderIlbcImpl(const CodecInst& codec_inst);
   ~AudioEncoderIlbcImpl() override;
 
   int SampleRateHz() const override;

modules/audio_coding/codecs/isac/audio_encoder_isac_t.h

@@ -20,8 +20,6 @@
 
 namespace webrtc {
 
-struct CodecInst;
-
 template <typename T>
 class AudioEncoderIsacT final : public AudioEncoder {
  public:
@@ -52,9 +50,6 @@ class AudioEncoderIsacT final : public AudioEncoder {
   };
 
   explicit AudioEncoderIsacT(const Config& config);
-  explicit AudioEncoderIsacT(
-      const CodecInst& codec_inst,
-      const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo);
   ~AudioEncoderIsacT() override;
 
   int SampleRateHz() const override;

modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h

@@ -11,27 +11,10 @@
 #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
 #define MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
 
-#include "common_types.h"  // NOLINT(build/include)
 #include "rtc_base/checks.h"
 
 namespace webrtc {
 
-template <typename T>
-typename AudioEncoderIsacT<T>::Config CreateIsacConfig(
-    const CodecInst& codec_inst,
-    const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo) {
-  typename AudioEncoderIsacT<T>::Config config;
-  config.bwinfo = bwinfo;
-  config.payload_type = codec_inst.pltype;
-  config.sample_rate_hz = codec_inst.plfreq;
-  config.frame_size_ms =
-      rtc::CheckedDivExact(1000 * codec_inst.pacsize, config.sample_rate_hz);
-  config.adaptive_mode = (codec_inst.rate == -1);
-  if (codec_inst.rate != -1)
-    config.bit_rate = codec_inst.rate;
-  return config;
-}
-
 template <typename T>
 bool AudioEncoderIsacT<T>::Config::IsOk() const {
   if (max_bit_rate < 32000 && max_bit_rate != -1)
@@ -66,12 +49,6 @@ AudioEncoderIsacT<T>::AudioEncoderIsacT(const Config& config) {
   RecreateEncoderInstance(config);
 }
 
-template <typename T>
-AudioEncoderIsacT<T>::AudioEncoderIsacT(
-    const CodecInst& codec_inst,
-    const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo)
-    : AudioEncoderIsacT(CreateIsacConfig<T>(codec_inst, bwinfo)) {}
-
 template <typename T>
 AudioEncoderIsacT<T>::~AudioEncoderIsacT() {
   RTC_CHECK_EQ(0, T::Free(isac_state_));

modules/audio_coding/codecs/opus/audio_encoder_opus.cc

@@ -16,7 +16,6 @@
 
 #include "absl/memory/memory.h"
 #include "absl/strings/match.h"
-#include "common_types.h"
 #include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
 #include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
 #include "modules/audio_coding/codecs/opus/opus_interface.h"
@@ -334,19 +333,6 @@ absl::optional<AudioCodecInfo> AudioEncoderOpusImpl::QueryAudioEncoder(
   return absl::nullopt;
 }
 
-AudioEncoderOpusConfig AudioEncoderOpusImpl::CreateConfig(
-    const CodecInst& codec_inst) {
-  AudioEncoderOpusConfig config;
-  config.frame_size_ms = rtc::CheckedDivExact(codec_inst.pacsize, 48);
-  config.num_channels = codec_inst.channels;
-  config.bitrate_bps = codec_inst.rate;
-  config.application = config.num_channels == 1
-                           ? AudioEncoderOpusConfig::ApplicationMode::kVoip
-                           : AudioEncoderOpusConfig::ApplicationMode::kAudio;
-  config.supported_frame_lengths_ms.push_back(config.frame_size_ms);
-  return config;
-}
-
 absl::optional<AudioEncoderOpusConfig> AudioEncoderOpusImpl::SdpToConfig(
     const SdpAudioFormat& format) {
   if (!absl::EqualsIgnoreCase(format.name, "opus") ||
@@ -494,9 +480,6 @@ AudioEncoderOpusImpl::AudioEncoderOpusImpl(
   SetProjectedPacketLossRate(packet_loss_rate_);
 }
 
-AudioEncoderOpusImpl::AudioEncoderOpusImpl(const CodecInst& codec_inst)
-    : AudioEncoderOpusImpl(CreateConfig(codec_inst), codec_inst.pltype) {}
-
 AudioEncoderOpusImpl::AudioEncoderOpusImpl(int payload_type,
                                            const SdpAudioFormat& format)
     : AudioEncoderOpusImpl(*SdpToConfig(format), payload_type) {}

modules/audio_coding/codecs/opus/audio_encoder_opus.h

@@ -30,8 +30,6 @@ namespace webrtc {
 
 class RtcEventLog;
 
-struct CodecInst;
-
 class AudioEncoderOpusImpl final : public AudioEncoder {
  public:
   class NewPacketLossRateOptimizer {
@@ -54,8 +52,6 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
     RTC_DISALLOW_COPY_AND_ASSIGN(NewPacketLossRateOptimizer);
   };
 
-  static AudioEncoderOpusConfig CreateConfig(const CodecInst& codec_inst);
-
   // Returns empty if the current bitrate falls within the hysteresis window,
   // defined by complexity_threshold_bps +/- complexity_threshold_window_bps.
   // Otherwise, returns the current complexity depending on whether the
@@ -83,7 +79,6 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
       const AudioNetworkAdaptorCreator& audio_network_adaptor_creator,
       std::unique_ptr<SmoothingFilter> bitrate_smoother);
 
-  explicit AudioEncoderOpusImpl(const CodecInst& codec_inst);
   AudioEncoderOpusImpl(int payload_type, const SdpAudioFormat& format);
   ~AudioEncoderOpusImpl() override;
 

modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc

@@ -15,7 +15,6 @@
 #include "absl/memory/memory.h"
 #include "api/audio_codecs/opus/audio_encoder_opus.h"
 #include "common_audio/mocks/mock_smoothing_filter.h"
-#include "common_types.h"  // NOLINT(build/include)
 #include "modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h"
 #include "modules/audio_coding/codecs/opus/audio_encoder_opus.h"
 #include "modules/audio_coding/codecs/opus/opus_interface.h"
@@ -33,21 +32,11 @@ using ::testing::Return;
 
 namespace {
 
-const CodecInst kDefaultOpusSettings = {105, "opus", 48000, 960, 1, 32000};
+constexpr int kDefaultOpusPayloadType = 105;
+constexpr int kDefaultOpusRate = 32000;
+constexpr int kDefaultOpusPacSize = 960;
 constexpr int64_t kInitialTimeUs = 12345678;
 
-AudioEncoderOpusConfig CreateConfig(const CodecInst& codec_inst) {
-  AudioEncoderOpusConfig config;
-  config.frame_size_ms = rtc::CheckedDivExact(codec_inst.pacsize, 48);
-  config.num_channels = codec_inst.channels;
-  config.bitrate_bps = codec_inst.rate;
-  config.application = config.num_channels == 1
-                           ? AudioEncoderOpusConfig::ApplicationMode::kVoip
-                           : AudioEncoderOpusConfig::ApplicationMode::kAudio;
-  config.supported_frame_lengths_ms.push_back(config.frame_size_ms);
-  return config;
-}
-
 AudioEncoderOpusConfig CreateConfigWithParameters(
     const SdpAudioFormat::Parameters& params) {
   const SdpAudioFormat format("opus", 48000, 2, params);
@@ -79,15 +68,22 @@ std::unique_ptr<AudioEncoderOpusStates> CreateCodec(size_t num_channels) {
         return adaptor;
       };
 
-  CodecInst codec_inst = kDefaultOpusSettings;
+  AudioEncoderOpusConfig config;
-  codec_inst.channels = num_channels;
+  config.frame_size_ms = rtc::CheckedDivExact(kDefaultOpusPacSize, 48);
-  states->config = CreateConfig(codec_inst);
+  config.num_channels = num_channels;
+  config.bitrate_bps = kDefaultOpusRate;
+  config.application = num_channels == 1
+                           ? AudioEncoderOpusConfig::ApplicationMode::kVoip
+                           : AudioEncoderOpusConfig::ApplicationMode::kAudio;
+  config.supported_frame_lengths_ms.push_back(config.frame_size_ms);
+  states->config = config;
+
   std::unique_ptr<MockSmoothingFilter> bitrate_smoother(
       new MockSmoothingFilter());
   states->mock_bitrate_smoother = bitrate_smoother.get();
 
   states->encoder.reset(new AudioEncoderOpusImpl(
-      states->config, codec_inst.pltype, std::move(creator),
+      states->config, kDefaultOpusPayloadType, std::move(creator),
       std::move(bitrate_smoother)));
   return states;
 }
@@ -436,12 +432,12 @@ TEST(AudioEncoderOpusTest, DoNotInvokeSetTargetBitrateIfOverheadUnknown) {
 
   auto states = CreateCodec(2);
 
-  states->encoder->OnReceivedUplinkBandwidth(kDefaultOpusSettings.rate * 2,
+  states->encoder->OnReceivedUplinkBandwidth(kDefaultOpusRate * 2,
                                              absl::nullopt);
 
   // Since |OnReceivedOverhead| has not been called, the codec bitrate should
   // not change.
-  EXPECT_EQ(kDefaultOpusSettings.rate, states->encoder->GetTargetBitrate());
+  EXPECT_EQ(kDefaultOpusRate, states->encoder->GetTargetBitrate());
 }
 
 TEST(AudioEncoderOpusTest, OverheadRemovedFromTargetAudioBitrate) {
@@ -456,7 +452,7 @@ TEST(AudioEncoderOpusTest, OverheadRemovedFromTargetAudioBitrate) {
   constexpr int kTargetBitrateBps = 40000;
   states->encoder->OnReceivedUplinkBandwidth(kTargetBitrateBps, absl::nullopt);
 
-  int packet_rate = rtc::CheckedDivExact(48000, kDefaultOpusSettings.pacsize);
+  int packet_rate = rtc::CheckedDivExact(48000, kDefaultOpusPacSize);
   EXPECT_EQ(kTargetBitrateBps -
                 8 * static_cast<int>(kOverheadBytesPerPacket) * packet_rate,
             states->encoder->GetTargetBitrate());
@@ -474,7 +470,7 @@ TEST(AudioEncoderOpusTest, BitrateBounded) {
   constexpr size_t kOverheadBytesPerPacket = 64;
   states->encoder->OnReceivedOverhead(kOverheadBytesPerPacket);
 
-  int packet_rate = rtc::CheckedDivExact(48000, kDefaultOpusSettings.pacsize);
+  int packet_rate = rtc::CheckedDivExact(48000, kDefaultOpusPacSize);
 
   // Set a target rate that is smaller than |kMinBitrateBps| when overhead is
   // subtracted. The eventual codec rate should be bounded by |kMinBitrateBps|.

modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc

@@ -10,7 +10,6 @@
 
 #include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
 
-#include "common_types.h"
 #include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "rtc_base/checks.h"
 
@@ -30,20 +29,6 @@ AudioEncoder::CodecType AudioEncoderPcm16B::GetCodecType() const {
   return CodecType::kOther;
 }
 
-namespace {
-
-AudioEncoderPcm16B::Config CreateConfig(const CodecInst& codec_inst) {
-  AudioEncoderPcm16B::Config config;
-  config.num_channels = codec_inst.channels;
-  config.sample_rate_hz = codec_inst.plfreq;
-  config.frame_size_ms = rtc::CheckedDivExact(
-      codec_inst.pacsize, rtc::CheckedDivExact(config.sample_rate_hz, 1000));
-  config.payload_type = codec_inst.pltype;
-  return config;
-}
-
-}  // namespace
-
 bool AudioEncoderPcm16B::Config::IsOk() const {
   if ((sample_rate_hz != 8000) && (sample_rate_hz != 16000) &&
       (sample_rate_hz != 32000) && (sample_rate_hz != 48000))
@@ -51,7 +36,4 @@ bool AudioEncoderPcm16B::Config::IsOk() const {
   return AudioEncoderPcm::Config::IsOk();
 }
 
-AudioEncoderPcm16B::AudioEncoderPcm16B(const CodecInst& codec_inst)
-    : AudioEncoderPcm16B(CreateConfig(codec_inst)) {}
-
 }  // namespace webrtc

modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h

@@ -16,8 +16,6 @@
 
 namespace webrtc {
 
-struct CodecInst;
-
 class AudioEncoderPcm16B final : public AudioEncoderPcm {
  public:
   struct Config : public AudioEncoderPcm::Config {
@@ -30,7 +28,6 @@ class AudioEncoderPcm16B final : public AudioEncoderPcm {
 
   explicit AudioEncoderPcm16B(const Config& config)
       : AudioEncoderPcm(config, config.sample_rate_hz) {}
-  explicit AudioEncoderPcm16B(const CodecInst& codec_inst);
 
  protected:
   size_t EncodeCall(const int16_t* audio,

modules/video_coding/utility/simulcast_rate_allocator_unittest.cc

@@ -16,6 +16,7 @@
 #include <vector>
 
 #include "api/video_codecs/vp8_temporal_layers.h"
+#include "rtc_base/checks.h"
 #include "test/field_trial.h"
 #include "test/gmock.h"
 #include "test/gtest.h"

pc/peerconnection.cc

@@ -18,6 +18,7 @@
 #include <vector>
 
 #include "absl/memory/memory.h"
+#include "absl/strings/match.h"
 #include "api/jsepicecandidate.h"
 #include "api/jsepsessiondescription.h"
 #include "api/mediastreamproxy.h"