Repo created

TMessagesProj/jni/voip/webrtc/api/audio_codecs/g722/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_source_set("audio_encoder_g722_config") {
+  visibility = [ "*" ]
+  sources = [ "audio_encoder_g722_config.h" ]
+  deps = [ "..:audio_codecs_api" ]
+}
+
+rtc_library("audio_encoder_g722") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_encoder_g722.cc",
+    "audio_encoder_g722.h",
+  ]
+  deps = [
+    ":audio_encoder_g722_config",
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:g722",
+    "../../../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_g722") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_decoder_g722.cc",
+    "audio_decoder_g722.h",
+  ]
+  deps = [
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:g722",
+    "../../../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/g722/audio_decoder_g722.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/g722/audio_decoder_g722.h"
+
+#include <memory>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/g722/audio_decoder_g722.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+absl::optional<AudioDecoderG722::Config> AudioDecoderG722::SdpToConfig(
+    const SdpAudioFormat& format) {
+  if (absl::EqualsIgnoreCase(format.name, "G722") &&
+      format.clockrate_hz == 8000 &&
+      (format.num_channels == 1 || format.num_channels == 2)) {
+    return Config{rtc::dchecked_cast<int>(format.num_channels)};
+  }
+  return absl::nullopt;
+}
+
+void AudioDecoderG722::AppendSupportedDecoders(
+    std::vector<AudioCodecSpec>* specs) {
+  specs->push_back({{"G722", 8000, 1}, {16000, 1, 64000}});
+}
+
+std::unique_ptr<AudioDecoder> AudioDecoderG722::MakeAudioDecoder(
+    Config config,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  switch (config.num_channels) {
+    case 1:
+      return std::make_unique<AudioDecoderG722Impl>();
+    case 2:
+      return std::make_unique<AudioDecoderG722StereoImpl>();
+    default:
+      RTC_DCHECK_NOTREACHED();
+      return nullptr;
+  }
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/g722/audio_decoder_g722.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_G722_AUDIO_DECODER_G722_H_
+#define API_AUDIO_CODECS_G722_AUDIO_DECODER_G722_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 {
+
+// G722 decoder API for use as a template parameter to
+// CreateAudioDecoderFactory<...>().
+struct RTC_EXPORT AudioDecoderG722 {
+  struct Config {
+    bool IsOk() const { return num_channels == 1 || num_channels == 2; }
+    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(
+      Config config,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_G722_AUDIO_DECODER_G722_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/g722/audio_encoder_g722.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/g722/audio_encoder_g722.h"
+
+#include <memory>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/g722/audio_encoder_g722.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<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig(
+    const SdpAudioFormat& format) {
+  if (!absl::EqualsIgnoreCase(format.name, "g722") ||
+      format.clockrate_hz != 8000) {
+    return absl::nullopt;
+  }
+
+  AudioEncoderG722Config config;
+  config.num_channels = rtc::checked_cast<int>(format.num_channels);
+  auto ptime_iter = format.parameters.find("ptime");
+  if (ptime_iter != format.parameters.end()) {
+    auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
+    if (ptime && *ptime > 0) {
+      const int whole_packets = *ptime / 10;
+      config.frame_size_ms = rtc::SafeClamp<int>(whole_packets * 10, 10, 60);
+    }
+  }
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return absl::nullopt;
+  }
+  return config;
+}
+
+void AudioEncoderG722::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  const SdpAudioFormat fmt = {"G722", 8000, 1};
+  const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
+  specs->push_back({fmt, info});
+}
+
+AudioCodecInfo AudioEncoderG722::QueryAudioEncoder(
+    const AudioEncoderG722Config& config) {
+  RTC_DCHECK(config.IsOk());
+  return {16000, rtc::dchecked_cast<size_t>(config.num_channels),
+          64000 * config.num_channels};
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderG722::MakeAudioEncoder(
+    const AudioEncoderG722Config& config,
+    int payload_type,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  return std::make_unique<AudioEncoderG722Impl>(config, payload_type);
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/g722/audio_encoder_g722.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_G722_AUDIO_ENCODER_G722_H_
+#define API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_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/g722/audio_encoder_g722_config.h"
+#include "api/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// G722 encoder API for use as a template parameter to
+// CreateAudioEncoderFactory<...>().
+struct RTC_EXPORT AudioEncoderG722 {
+  using Config = AudioEncoderG722Config;
+  static absl::optional<AudioEncoderG722Config> SdpToConfig(
+      const SdpAudioFormat& audio_format);
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+  static AudioCodecInfo QueryAudioEncoder(const AudioEncoderG722Config& config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const AudioEncoderG722Config& config,
+      int payload_type,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/g722/audio_encoder_g722_config.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_G722_AUDIO_ENCODER_G722_CONFIG_H_
+#define API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_CONFIG_H_
+
+#include "api/audio_codecs/audio_encoder.h"
+
+namespace webrtc {
+
+struct AudioEncoderG722Config {
+  bool IsOk() const {
+    return frame_size_ms > 0 && frame_size_ms % 10 == 0 && num_channels >= 1 &&
+           num_channels <= AudioEncoder::kMaxNumberOfChannels;
+  }
+  int frame_size_ms = 20;
+  int num_channels = 1;
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_CONFIG_H_