Repo created

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/BUILD.gn

@@ -0,0 +1,110 @@
+# Copyright (c) 2017 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.
+
+import("../../../webrtc.gni")
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
+
+rtc_library("audio_encoder_opus_config") {
+  visibility = [ "*" ]
+  sources = [
+    "audio_encoder_multi_channel_opus_config.cc",
+    "audio_encoder_multi_channel_opus_config.h",
+    "audio_encoder_opus_config.cc",
+    "audio_encoder_opus_config.h",
+  ]
+  deps = [ "../../../rtc_base/system:rtc_export" ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+  defines = []
+  if (rtc_opus_variable_complexity) {
+    defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=1" ]
+  } else {
+    defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=0" ]
+  }
+}
+
+rtc_source_set("audio_decoder_opus_config") {
+  visibility = [ "*" ]
+  sources = [ "audio_decoder_multi_channel_opus_config.h" ]
+  deps = [ "..:audio_codecs_api" ]
+}
+
+rtc_library("audio_encoder_opus") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  public = [ "audio_encoder_opus.h" ]
+  sources = [ "audio_encoder_opus.cc" ]
+  deps = [
+    ":audio_encoder_opus_config",
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:webrtc_opus",
+    "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("audio_decoder_opus") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_decoder_opus.cc",
+    "audio_decoder_opus.h",
+  ]
+  deps = [
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:webrtc_opus",
+    "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("audio_encoder_multiopus") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  public = [ "audio_encoder_multi_channel_opus.h" ]
+  sources = [ "audio_encoder_multi_channel_opus.cc" ]
+  deps = [
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:webrtc_multiopus",
+    "../../../rtc_base/system:rtc_export",
+    "../opus:audio_encoder_opus_config",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("audio_decoder_multiopus") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_decoder_multi_channel_opus.cc",
+    "audio_decoder_multi_channel_opus.h",
+  ]
+  deps = [
+    ":audio_decoder_opus_config",
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:webrtc_multiopus",
+    "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus.cc

@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2019 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/opus/audio_decoder_multi_channel_opus.h"
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h"
+
+namespace webrtc {
+
+absl::optional<AudioDecoderMultiChannelOpusConfig>
+AudioDecoderMultiChannelOpus::SdpToConfig(const SdpAudioFormat& format) {
+  return AudioDecoderMultiChannelOpusImpl::SdpToConfig(format);
+}
+
+void AudioDecoderMultiChannelOpus::AppendSupportedDecoders(
+    std::vector<AudioCodecSpec>* specs) {
+  // To get full utilization of the surround support of the Opus lib, we can
+  // mark which channel is the low frequency effects (LFE). But that is not done
+  // ATM.
+  {
+    AudioCodecInfo surround_5_1_opus_info{48000, 6,
+                                          /* default_bitrate_bps= */ 128000};
+    surround_5_1_opus_info.allow_comfort_noise = false;
+    surround_5_1_opus_info.supports_network_adaption = false;
+    SdpAudioFormat opus_format({"multiopus",
+                                48000,
+                                6,
+                                {{"minptime", "10"},
+                                 {"useinbandfec", "1"},
+                                 {"channel_mapping", "0,4,1,2,3,5"},
+                                 {"num_streams", "4"},
+                                 {"coupled_streams", "2"}}});
+    specs->push_back({std::move(opus_format), surround_5_1_opus_info});
+  }
+  {
+    AudioCodecInfo surround_7_1_opus_info{48000, 8,
+                                          /* default_bitrate_bps= */ 200000};
+    surround_7_1_opus_info.allow_comfort_noise = false;
+    surround_7_1_opus_info.supports_network_adaption = false;
+    SdpAudioFormat opus_format({"multiopus",
+                                48000,
+                                8,
+                                {{"minptime", "10"},
+                                 {"useinbandfec", "1"},
+                                 {"channel_mapping", "0,6,1,2,3,4,5,7"},
+                                 {"num_streams", "5"},
+                                 {"coupled_streams", "3"}}});
+    specs->push_back({std::move(opus_format), surround_7_1_opus_info});
+  }
+}
+
+std::unique_ptr<AudioDecoder> AudioDecoderMultiChannelOpus::MakeAudioDecoder(
+    AudioDecoderMultiChannelOpusConfig config,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  return AudioDecoderMultiChannelOpusImpl::MakeAudioDecoder(config);
+}
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus.h

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+/*
+ *  Copyright (c) 2019 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 API_AUDIO_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_H_
+#define API_AUDIO_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h"
+#include "api/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// Opus decoder API for use as a template parameter to
+// CreateAudioDecoderFactory<...>().
+struct RTC_EXPORT AudioDecoderMultiChannelOpus {
+  using Config = AudioDecoderMultiChannelOpusConfig;
+  static absl::optional<AudioDecoderMultiChannelOpusConfig> SdpToConfig(
+      const SdpAudioFormat& audio_format);
+  static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      AudioDecoderMultiChannelOpusConfig config,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h

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+/*
+ *  Copyright (c) 2019 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 API_AUDIO_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_CONFIG_H_
+#define API_AUDIO_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_CONFIG_H_
+
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder.h"
+
+namespace webrtc {
+struct AudioDecoderMultiChannelOpusConfig {
+  // The number of channels that the decoder will output.
+  int num_channels;
+
+  // Number of mono or stereo encoded Opus streams.
+  int num_streams;
+
+  // Number of channel pairs coupled together, see RFC 7845 section
+  // 5.1.1. Has to be less than the number of streams.
+  int coupled_streams;
+
+  // Channel mapping table, defines the mapping from encoded streams to output
+  // channels. See RFC 7845 section 5.1.1.
+  std::vector<unsigned char> channel_mapping;
+
+  bool IsOk() const {
+    if (num_channels < 1 || num_channels > AudioDecoder::kMaxNumberOfChannels ||
+        num_streams < 0 || coupled_streams < 0) {
+      return false;
+    }
+    if (num_streams < coupled_streams) {
+      return false;
+    }
+    if (channel_mapping.size() != static_cast<size_t>(num_channels)) {
+      return false;
+    }
+
+    // Every mono stream codes one channel, every coupled stream codes two. This
+    // is the total coded channel count:
+    const int max_coded_channel = num_streams + coupled_streams;
+    for (const auto& x : channel_mapping) {
+      // Coded channels >= max_coded_channel don't exist. Except for 255, which
+      // tells Opus to put silence in output channel x.
+      if (x >= max_coded_channel && x != 255) {
+        return false;
+      }
+    }
+
+    if (num_channels > 255 || max_coded_channel >= 255) {
+      return false;
+    }
+    return true;
+  }
+};
+
+}  // namespace webrtc
+
+#endif  //  API_AUDIO_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_CONFIG_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/audio_decoder_opus.cc

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+/*
+ *  Copyright (c) 2017 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/opus/audio_decoder_opus.h"
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
+
+namespace webrtc {
+
+bool AudioDecoderOpus::Config::IsOk() const {
+  if (sample_rate_hz != 16000 && sample_rate_hz != 48000) {
+    // Unsupported sample rate. (libopus supports a few other rates as
+    // well; we can add support for them when needed.)
+    return false;
+  }
+  if (num_channels != 1 && num_channels != 2) {
+    return false;
+  }
+  return true;
+}
+
+absl::optional<AudioDecoderOpus::Config> AudioDecoderOpus::SdpToConfig(
+    const SdpAudioFormat& format) {
+  const auto num_channels = [&]() -> absl::optional<int> {
+    auto stereo = format.parameters.find("stereo");
+    if (stereo != format.parameters.end()) {
+      if (stereo->second == "0") {
+        return 1;
+      } else if (stereo->second == "1") {
+        return 2;
+      } else {
+        return absl::nullopt;  // Bad stereo parameter.
+      }
+    }
+    return 1;  // Default to mono.
+  }();
+  if (absl::EqualsIgnoreCase(format.name, "opus") &&
+      format.clockrate_hz == 48000 && format.num_channels == 2 &&
+      num_channels) {
+    Config config;
+    config.num_channels = *num_channels;
+    if (!config.IsOk()) {
+      RTC_DCHECK_NOTREACHED();
+      return absl::nullopt;
+    }
+    return config;
+  } else {
+    return absl::nullopt;
+  }
+}
+
+void AudioDecoderOpus::AppendSupportedDecoders(
+    std::vector<AudioCodecSpec>* specs) {
+  AudioCodecInfo opus_info{48000, 1, 64000, 6000, 510000};
+  opus_info.allow_comfort_noise = false;
+  opus_info.supports_network_adaption = true;
+  SdpAudioFormat opus_format(
+      {"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}});
+  specs->push_back({std::move(opus_format), opus_info});
+}
+
+std::unique_ptr<AudioDecoder> AudioDecoderOpus::MakeAudioDecoder(
+    Config config,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  return std::make_unique<AudioDecoderOpusImpl>(config.num_channels,
+                                                config.sample_rate_hz);
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/audio_decoder_opus.h

@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2017 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 API_AUDIO_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
+#define API_AUDIO_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// Opus decoder API for use as a template parameter to
+// CreateAudioDecoderFactory<...>().
+struct RTC_EXPORT AudioDecoderOpus {
+  struct Config {
+    bool IsOk() const;  // Checks if the values are currently OK.
+    int sample_rate_hz = 48000;
+    int num_channels = 1;
+  };
+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+  static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      Config config,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_OPUS_AUDIO_DECODER_OPUS_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus.cc

@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2019 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/opus/audio_encoder_multi_channel_opus.h"
+
+#include <utility>
+
+#include "modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h"
+
+namespace webrtc {
+
+absl::optional<AudioEncoderMultiChannelOpusConfig>
+AudioEncoderMultiChannelOpus::SdpToConfig(const SdpAudioFormat& format) {
+  return AudioEncoderMultiChannelOpusImpl::SdpToConfig(format);
+}
+
+void AudioEncoderMultiChannelOpus::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  // To get full utilization of the surround support of the Opus lib, we can
+  // mark which channel is the low frequency effects (LFE). But that is not done
+  // ATM.
+  {
+    AudioCodecInfo surround_5_1_opus_info{48000, 6,
+                                          /* default_bitrate_bps= */ 128000};
+    surround_5_1_opus_info.allow_comfort_noise = false;
+    surround_5_1_opus_info.supports_network_adaption = false;
+    SdpAudioFormat opus_format({"multiopus",
+                                48000,
+                                6,
+                                {{"minptime", "10"},
+                                 {"useinbandfec", "1"},
+                                 {"channel_mapping", "0,4,1,2,3,5"},
+                                 {"num_streams", "4"},
+                                 {"coupled_streams", "2"}}});
+    specs->push_back({std::move(opus_format), surround_5_1_opus_info});
+  }
+  {
+    AudioCodecInfo surround_7_1_opus_info{48000, 8,
+                                          /* default_bitrate_bps= */ 200000};
+    surround_7_1_opus_info.allow_comfort_noise = false;
+    surround_7_1_opus_info.supports_network_adaption = false;
+    SdpAudioFormat opus_format({"multiopus",
+                                48000,
+                                8,
+                                {{"minptime", "10"},
+                                 {"useinbandfec", "1"},
+                                 {"channel_mapping", "0,6,1,2,3,4,5,7"},
+                                 {"num_streams", "5"},
+                                 {"coupled_streams", "3"}}});
+    specs->push_back({std::move(opus_format), surround_7_1_opus_info});
+  }
+}
+
+AudioCodecInfo AudioEncoderMultiChannelOpus::QueryAudioEncoder(
+    const AudioEncoderMultiChannelOpusConfig& config) {
+  return AudioEncoderMultiChannelOpusImpl::QueryAudioEncoder(config);
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderMultiChannelOpus::MakeAudioEncoder(
+    const AudioEncoderMultiChannelOpusConfig& config,
+    int payload_type,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  return AudioEncoderMultiChannelOpusImpl::MakeAudioEncoder(config,
+                                                            payload_type);
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus.h

@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2019 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 API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_H_
+#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h"
+#include "api/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// Opus encoder API for use as a template parameter to
+// CreateAudioEncoderFactory<...>().
+struct RTC_EXPORT AudioEncoderMultiChannelOpus {
+  using Config = AudioEncoderMultiChannelOpusConfig;
+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+  static AudioCodecInfo QueryAudioEncoder(const Config& config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const Config& config,
+      int payload_type,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.cc

@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2019 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/opus/audio_encoder_multi_channel_opus_config.h"
+
+namespace webrtc {
+
+namespace {
+constexpr int kDefaultComplexity = 9;
+}  // namespace
+
+AudioEncoderMultiChannelOpusConfig::AudioEncoderMultiChannelOpusConfig()
+    : frame_size_ms(kDefaultFrameSizeMs),
+      num_channels(1),
+      application(ApplicationMode::kVoip),
+      bitrate_bps(32000),
+      fec_enabled(false),
+      cbr_enabled(false),
+      dtx_enabled(false),
+      max_playback_rate_hz(48000),
+      complexity(kDefaultComplexity),
+      num_streams(-1),
+      coupled_streams(-1) {}
+AudioEncoderMultiChannelOpusConfig::AudioEncoderMultiChannelOpusConfig(
+    const AudioEncoderMultiChannelOpusConfig&) = default;
+AudioEncoderMultiChannelOpusConfig::~AudioEncoderMultiChannelOpusConfig() =
+    default;
+AudioEncoderMultiChannelOpusConfig&
+AudioEncoderMultiChannelOpusConfig::operator=(
+    const AudioEncoderMultiChannelOpusConfig&) = default;
+
+bool AudioEncoderMultiChannelOpusConfig::IsOk() const {
+  if (frame_size_ms <= 0 || frame_size_ms % 10 != 0)
+    return false;
+  if (num_channels >= 255) {
+    return false;
+  }
+  if (bitrate_bps < kMinBitrateBps || bitrate_bps > kMaxBitrateBps)
+    return false;
+  if (complexity < 0 || complexity > 10)
+    return false;
+
+  // Check the lengths:
+  if (num_streams < 0 || coupled_streams < 0) {
+    return false;
+  }
+  if (num_streams < coupled_streams) {
+    return false;
+  }
+  if (channel_mapping.size() != static_cast<size_t>(num_channels)) {
+    return false;
+  }
+
+  // Every mono stream codes one channel, every coupled stream codes two. This
+  // is the total coded channel count:
+  const int max_coded_channel = num_streams + coupled_streams;
+  for (const auto& x : channel_mapping) {
+    // Coded channels >= max_coded_channel don't exist. Except for 255, which
+    // tells Opus to ignore input channel x.
+    if (x >= max_coded_channel && x != 255) {
+      return false;
+    }
+  }
+
+  // Inverse mapping.
+  constexpr int kNotSet = -1;
+  std::vector<int> coded_channels_to_input_channels(max_coded_channel, kNotSet);
+  for (size_t i = 0; i < num_channels; ++i) {
+    if (channel_mapping[i] == 255) {
+      continue;
+    }
+
+    // If it's not ignored, put it in the inverted mapping. But first check if
+    // we've told Opus to use another input channel for this coded channel:
+    const int coded_channel = channel_mapping[i];
+    if (coded_channels_to_input_channels[coded_channel] != kNotSet) {
+      // Coded channel `coded_channel` comes from both input channels
+      // `coded_channels_to_input_channels[coded_channel]` and `i`.
+      return false;
+    }
+
+    coded_channels_to_input_channels[coded_channel] = i;
+  }
+
+  // Check that we specified what input the encoder should use to produce
+  // every coded channel.
+  for (int i = 0; i < max_coded_channel; ++i) {
+    if (coded_channels_to_input_channels[i] == kNotSet) {
+      // Coded channel `i` has unspecified input channel.
+      return false;
+    }
+  }
+
+  if (num_channels > 255 || max_coded_channel >= 255) {
+    return false;
+  }
+  return true;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h

@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2019 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 API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_CONFIG_H_
+#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_CONFIG_H_
+
+#include <stddef.h>
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/opus/audio_encoder_opus_config.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+struct RTC_EXPORT AudioEncoderMultiChannelOpusConfig {
+  static constexpr int kDefaultFrameSizeMs = 20;
+
+  // Opus API allows a min bitrate of 500bps, but Opus documentation suggests
+  // bitrate should be in the range of 6000 to 510000, inclusive.
+  static constexpr int kMinBitrateBps = 6000;
+  static constexpr int kMaxBitrateBps = 510000;
+
+  AudioEncoderMultiChannelOpusConfig();
+  AudioEncoderMultiChannelOpusConfig(const AudioEncoderMultiChannelOpusConfig&);
+  ~AudioEncoderMultiChannelOpusConfig();
+  AudioEncoderMultiChannelOpusConfig& operator=(
+      const AudioEncoderMultiChannelOpusConfig&);
+
+  int frame_size_ms;
+  size_t num_channels;
+  enum class ApplicationMode { kVoip, kAudio };
+  ApplicationMode application;
+  int bitrate_bps;
+  bool fec_enabled;
+  bool cbr_enabled;
+  bool dtx_enabled;
+  int max_playback_rate_hz;
+  std::vector<int> supported_frame_lengths_ms;
+
+  int complexity;
+
+  // Number of mono/stereo Opus streams.
+  int num_streams;
+
+  // Number of channel pairs coupled together, see RFC 7845 section
+  // 5.1.1. Has to be less than the number of streams
+  int coupled_streams;
+
+  // Channel mapping table, defines the mapping from encoded streams to input
+  // channels. See RFC 7845 section 5.1.1.
+  std::vector<unsigned char> channel_mapping;
+
+  bool IsOk() const;
+};
+
+}  // namespace webrtc
+#endif  // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_CONFIG_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/audio_encoder_opus.cc

@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2017 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/opus/audio_encoder_opus.h"
+
+#include "modules/audio_coding/codecs/opus/audio_encoder_opus.h"
+
+namespace webrtc {
+
+absl::optional<AudioEncoderOpusConfig> AudioEncoderOpus::SdpToConfig(
+    const SdpAudioFormat& format) {
+  return AudioEncoderOpusImpl::SdpToConfig(format);
+}
+
+void AudioEncoderOpus::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  AudioEncoderOpusImpl::AppendSupportedEncoders(specs);
+}
+
+AudioCodecInfo AudioEncoderOpus::QueryAudioEncoder(
+    const AudioEncoderOpusConfig& config) {
+  return AudioEncoderOpusImpl::QueryAudioEncoder(config);
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderOpus::MakeAudioEncoder(
+    const AudioEncoderOpusConfig& config,
+    int payload_type,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  return AudioEncoderOpusImpl::MakeAudioEncoder(config, payload_type);
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/audio_encoder_opus.h

@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2017 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 API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
+#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/opus/audio_encoder_opus_config.h"
+#include "api/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// Opus encoder API for use as a template parameter to
+// CreateAudioEncoderFactory<...>().
+struct RTC_EXPORT AudioEncoderOpus {
+  using Config = AudioEncoderOpusConfig;
+  static absl::optional<AudioEncoderOpusConfig> SdpToConfig(
+      const SdpAudioFormat& audio_format);
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+  static AudioCodecInfo QueryAudioEncoder(const AudioEncoderOpusConfig& config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const AudioEncoderOpusConfig& config,
+      int payload_type,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc

@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2017 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/opus/audio_encoder_opus_config.h"
+
+namespace webrtc {
+
+namespace {
+
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
+constexpr int kDefaultComplexity = 5;
+#else
+constexpr int kDefaultComplexity = 9;
+#endif
+
+constexpr int kDefaultLowRateComplexity =
+    WEBRTC_OPUS_VARIABLE_COMPLEXITY ? 9 : kDefaultComplexity;
+
+}  // namespace
+
+constexpr int AudioEncoderOpusConfig::kDefaultFrameSizeMs;
+constexpr int AudioEncoderOpusConfig::kMinBitrateBps;
+constexpr int AudioEncoderOpusConfig::kMaxBitrateBps;
+
+AudioEncoderOpusConfig::AudioEncoderOpusConfig()
+    : frame_size_ms(kDefaultFrameSizeMs),
+      sample_rate_hz(48000),
+      num_channels(1),
+      application(ApplicationMode::kVoip),
+      bitrate_bps(32000),
+      fec_enabled(false),
+      cbr_enabled(false),
+      max_playback_rate_hz(48000),
+      complexity(kDefaultComplexity),
+      low_rate_complexity(kDefaultLowRateComplexity),
+      complexity_threshold_bps(12500),
+      complexity_threshold_window_bps(1500),
+      dtx_enabled(false),
+      uplink_bandwidth_update_interval_ms(200),
+      payload_type(-1) {}
+AudioEncoderOpusConfig::AudioEncoderOpusConfig(const AudioEncoderOpusConfig&) =
+    default;
+AudioEncoderOpusConfig::~AudioEncoderOpusConfig() = default;
+AudioEncoderOpusConfig& AudioEncoderOpusConfig::operator=(
+    const AudioEncoderOpusConfig&) = default;
+
+bool AudioEncoderOpusConfig::IsOk() const {
+  if (frame_size_ms <= 0 || frame_size_ms % 10 != 0)
+    return false;
+  if (sample_rate_hz != 16000 && sample_rate_hz != 48000) {
+    // Unsupported input sample rate. (libopus supports a few other rates as
+    // well; we can add support for them when needed.)
+    return false;
+  }
+  if (num_channels >= 255) {
+    return false;
+  }
+  if (!bitrate_bps)
+    return false;
+  if (*bitrate_bps < kMinBitrateBps || *bitrate_bps > kMaxBitrateBps)
+    return false;
+  if (complexity < 0 || complexity > 10)
+    return false;
+  if (low_rate_complexity < 0 || low_rate_complexity > 10)
+    return false;
+  return true;
+}
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h

@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2017 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 API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_
+#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_
+
+#include <stddef.h>
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+struct RTC_EXPORT AudioEncoderOpusConfig {
+  static constexpr int kDefaultFrameSizeMs = 20;
+
+  // Opus API allows a min bitrate of 500bps, but Opus documentation suggests
+  // bitrate should be in the range of 6000 to 510000, inclusive.
+  static constexpr int kMinBitrateBps = 6000;
+  static constexpr int kMaxBitrateBps = 510000;
+
+  AudioEncoderOpusConfig();
+  AudioEncoderOpusConfig(const AudioEncoderOpusConfig&);
+  ~AudioEncoderOpusConfig();
+  AudioEncoderOpusConfig& operator=(const AudioEncoderOpusConfig&);
+
+  bool IsOk() const;  // Checks if the values are currently OK.
+
+  int frame_size_ms;
+  int sample_rate_hz;
+  size_t num_channels;
+  enum class ApplicationMode { kVoip, kAudio };
+  ApplicationMode application;
+
+  // NOTE: This member must always be set.
+  // TODO(kwiberg): Turn it into just an int.
+  absl::optional<int> bitrate_bps;
+
+  bool fec_enabled;
+  bool cbr_enabled;
+  int max_playback_rate_hz;
+
+  // `complexity` is used when the bitrate goes above
+  // `complexity_threshold_bps` + `complexity_threshold_window_bps`;
+  // `low_rate_complexity` is used when the bitrate falls below
+  // `complexity_threshold_bps` - `complexity_threshold_window_bps`. In the
+  // interval in the middle, we keep using the most recent of the two
+  // complexity settings.
+  int complexity;
+  int low_rate_complexity;
+  int complexity_threshold_bps;
+  int complexity_threshold_window_bps;
+
+  bool dtx_enabled;
+  std::vector<int> supported_frame_lengths_ms;
+  int uplink_bandwidth_update_interval_ms;
+
+  // NOTE: This member isn't necessary, and will soon go away. See
+  // https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
+  int payload_type;
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_