Repo created

TMessagesProj/jni/voip/webrtc/api/audio_codecs/g711/BUILD.gn

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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.
+
+import("../../../webrtc.gni")
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
+
+rtc_library("audio_encoder_g711") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_encoder_g711.cc",
+    "audio_encoder_g711.h",
+  ]
+  deps = [
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:g711",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base:safe_minmax",
+    "../../../rtc_base:stringutils",
+    "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("audio_decoder_g711") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_decoder_g711.cc",
+    "audio_decoder_g711.h",
+  ]
+  deps = [
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:g711",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}

TMessagesProj/jni/voip/webrtc/api/audio_codecs/g711/audio_decoder_g711.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/g711/audio_decoder_g711.h"
+
+#include <memory>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+absl::optional<AudioDecoderG711::Config> AudioDecoderG711::SdpToConfig(
+    const SdpAudioFormat& format) {
+  const bool is_pcmu = absl::EqualsIgnoreCase(format.name, "PCMU");
+  const bool is_pcma = absl::EqualsIgnoreCase(format.name, "PCMA");
+  if (format.clockrate_hz == 8000 && format.num_channels >= 1 &&
+      (is_pcmu || is_pcma)) {
+    Config config;
+    config.type = is_pcmu ? Config::Type::kPcmU : Config::Type::kPcmA;
+    config.num_channels = rtc::dchecked_cast<int>(format.num_channels);
+    if (!config.IsOk()) {
+      RTC_DCHECK_NOTREACHED();
+      return absl::nullopt;
+    }
+    return config;
+  } else {
+    return absl::nullopt;
+  }
+}
+
+void AudioDecoderG711::AppendSupportedDecoders(
+    std::vector<AudioCodecSpec>* specs) {
+  for (const char* type : {"PCMU", "PCMA"}) {
+    specs->push_back({{type, 8000, 1}, {8000, 1, 64000}});
+  }
+}
+
+std::unique_ptr<AudioDecoder> AudioDecoderG711::MakeAudioDecoder(
+    const Config& config,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  switch (config.type) {
+    case Config::Type::kPcmU:
+      return std::make_unique<AudioDecoderPcmU>(config.num_channels);
+    case Config::Type::kPcmA:
+      return std::make_unique<AudioDecoderPcmA>(config.num_channels);
+    default:
+      RTC_DCHECK_NOTREACHED();
+      return nullptr;
+  }
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/g711/audio_decoder_g711.h

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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.
+ */
+
+#ifndef API_AUDIO_CODECS_G711_AUDIO_DECODER_G711_H_
+#define API_AUDIO_CODECS_G711_AUDIO_DECODER_G711_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 {
+
+// G711 decoder API for use as a template parameter to
+// CreateAudioDecoderFactory<...>().
+struct RTC_EXPORT AudioDecoderG711 {
+  struct Config {
+    enum class Type { kPcmU, kPcmA };
+    bool IsOk() const {
+      return (type == Type::kPcmU || type == Type::kPcmA) &&
+             num_channels >= 1 &&
+             num_channels <= AudioDecoder::kMaxNumberOfChannels;
+    }
+    Type type;
+    int num_channels;
+  };
+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+  static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      const Config& config,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_G711_AUDIO_DECODER_G711_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/g711/audio_encoder_g711.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/g711/audio_encoder_g711.h"
+
+#include <memory>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/string_to_number.h"
+
+namespace webrtc {
+
+absl::optional<AudioEncoderG711::Config> AudioEncoderG711::SdpToConfig(
+    const SdpAudioFormat& format) {
+  const bool is_pcmu = absl::EqualsIgnoreCase(format.name, "PCMU");
+  const bool is_pcma = absl::EqualsIgnoreCase(format.name, "PCMA");
+  if (format.clockrate_hz == 8000 && format.num_channels >= 1 &&
+      (is_pcmu || is_pcma)) {
+    Config config;
+    config.type = is_pcmu ? Config::Type::kPcmU : Config::Type::kPcmA;
+    config.num_channels = rtc::dchecked_cast<int>(format.num_channels);
+    config.frame_size_ms = 20;
+    auto ptime_iter = format.parameters.find("ptime");
+    if (ptime_iter != format.parameters.end()) {
+      const auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
+      if (ptime && *ptime > 0) {
+        config.frame_size_ms = rtc::SafeClamp(10 * (*ptime / 10), 10, 60);
+      }
+    }
+    if (!config.IsOk()) {
+      RTC_DCHECK_NOTREACHED();
+      return absl::nullopt;
+    }
+    return config;
+  } else {
+    return absl::nullopt;
+  }
+}
+
+void AudioEncoderG711::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  for (const char* type : {"PCMU", "PCMA"}) {
+    specs->push_back({{type, 8000, 1}, {8000, 1, 64000}});
+  }
+}
+
+AudioCodecInfo AudioEncoderG711::QueryAudioEncoder(const Config& config) {
+  RTC_DCHECK(config.IsOk());
+  return {8000, rtc::dchecked_cast<size_t>(config.num_channels),
+          64000 * config.num_channels};
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderG711::MakeAudioEncoder(
+    const Config& config,
+    int payload_type,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  switch (config.type) {
+    case Config::Type::kPcmU: {
+      AudioEncoderPcmU::Config impl_config;
+      impl_config.num_channels = config.num_channels;
+      impl_config.frame_size_ms = config.frame_size_ms;
+      impl_config.payload_type = payload_type;
+      return std::make_unique<AudioEncoderPcmU>(impl_config);
+    }
+    case Config::Type::kPcmA: {
+      AudioEncoderPcmA::Config impl_config;
+      impl_config.num_channels = config.num_channels;
+      impl_config.frame_size_ms = config.frame_size_ms;
+      impl_config.payload_type = payload_type;
+      return std::make_unique<AudioEncoderPcmA>(impl_config);
+    }
+    default: {
+      RTC_DCHECK_NOTREACHED();
+      return nullptr;
+    }
+  }
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/g711/audio_encoder_g711.h

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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.
+ */
+
+#ifndef API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_H_
+#define API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_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/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// G711 encoder API for use as a template parameter to
+// CreateAudioEncoderFactory<...>().
+struct RTC_EXPORT AudioEncoderG711 {
+  struct Config {
+    enum class Type { kPcmU, kPcmA };
+    bool IsOk() const {
+      return (type == Type::kPcmU || type == Type::kPcmA) &&
+             frame_size_ms > 0 && frame_size_ms % 10 == 0 &&
+             num_channels >= 1 &&
+             num_channels <= AudioEncoder::kMaxNumberOfChannels;
+    }
+    Type type = Type::kPcmU;
+    int num_channels = 1;
+    int frame_size_ms = 20;
+  };
+  static absl::optional<AudioEncoderG711::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_G711_AUDIO_ENCODER_G711_H_