Repo created

TMessagesProj/jni/voip/webrtc/api/audio_codecs/ilbc/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_ilbc_config") {
+  visibility = [ "*" ]
+  sources = [ "audio_encoder_ilbc_config.h" ]
+}
+
+rtc_library("audio_encoder_ilbc") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_encoder_ilbc.cc",
+    "audio_encoder_ilbc.h",
+  ]
+  deps = [
+    ":audio_encoder_ilbc_config",
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:ilbc",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base:safe_minmax",
+    "../../../rtc_base:stringutils",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("audio_decoder_ilbc") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_decoder_ilbc.cc",
+    "audio_decoder_ilbc.h",
+  ]
+  deps = [
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:ilbc",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}

TMessagesProj/jni/voip/webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.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/ilbc/audio_decoder_ilbc.h"
+
+#include <memory>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
+
+namespace webrtc {
+
+absl::optional<AudioDecoderIlbc::Config> AudioDecoderIlbc::SdpToConfig(
+    const SdpAudioFormat& format) {
+  if (absl::EqualsIgnoreCase(format.name, "ILBC") &&
+      format.clockrate_hz == 8000 && format.num_channels == 1) {
+    return Config();
+  }
+  return absl::nullopt;
+}
+
+void AudioDecoderIlbc::AppendSupportedDecoders(
+    std::vector<AudioCodecSpec>* specs) {
+  specs->push_back({{"ILBC", 8000, 1}, {8000, 1, 13300}});
+}
+
+std::unique_ptr<AudioDecoder> AudioDecoderIlbc::MakeAudioDecoder(
+    Config config,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  return std::make_unique<AudioDecoderIlbcImpl>();
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.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_ILBC_AUDIO_DECODER_ILBC_H_
+#define API_AUDIO_CODECS_ILBC_AUDIO_DECODER_ILBC_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"
+
+namespace webrtc {
+
+// ILBC decoder API for use as a template parameter to
+// CreateAudioDecoderFactory<...>().
+struct AudioDecoderIlbc {
+  struct Config {};  // Empty---no config values needed!
+  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_ILBC_AUDIO_DECODER_ILBC_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.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/ilbc/audio_encoder_ilbc.h"
+
+#include <memory>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/string_to_number.h"
+
+namespace webrtc {
+namespace {
+int GetIlbcBitrate(int ptime) {
+  switch (ptime) {
+    case 20:
+    case 40:
+      // 38 bytes per frame of 20 ms => 15200 bits/s.
+      return 15200;
+    case 30:
+    case 60:
+      // 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
+      return 13333;
+    default:
+      RTC_CHECK_NOTREACHED();
+  }
+}
+}  // namespace
+
+absl::optional<AudioEncoderIlbcConfig> AudioEncoderIlbc::SdpToConfig(
+    const SdpAudioFormat& format) {
+  if (!absl::EqualsIgnoreCase(format.name.c_str(), "ILBC") ||
+      format.clockrate_hz != 8000 || format.num_channels != 1) {
+    return absl::nullopt;
+  }
+
+  AudioEncoderIlbcConfig config;
+  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, 20, 60);
+    }
+  }
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return absl::nullopt;
+  }
+  return config;
+}
+
+void AudioEncoderIlbc::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  const SdpAudioFormat fmt = {"ILBC", 8000, 1};
+  const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
+  specs->push_back({fmt, info});
+}
+
+AudioCodecInfo AudioEncoderIlbc::QueryAudioEncoder(
+    const AudioEncoderIlbcConfig& config) {
+  RTC_DCHECK(config.IsOk());
+  return {8000, 1, GetIlbcBitrate(config.frame_size_ms)};
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderIlbc::MakeAudioEncoder(
+    const AudioEncoderIlbcConfig& 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<AudioEncoderIlbcImpl>(config, payload_type);
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.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_ILBC_AUDIO_ENCODER_ILBC_H_
+#define API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_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/ilbc/audio_encoder_ilbc_config.h"
+#include "api/field_trials_view.h"
+
+namespace webrtc {
+
+// ILBC encoder API for use as a template parameter to
+// CreateAudioEncoderFactory<...>().
+struct AudioEncoderIlbc {
+  using Config = AudioEncoderIlbcConfig;
+  static absl::optional<AudioEncoderIlbcConfig> SdpToConfig(
+      const SdpAudioFormat& audio_format);
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+  static AudioCodecInfo QueryAudioEncoder(const AudioEncoderIlbcConfig& config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const AudioEncoderIlbcConfig& config,
+      int payload_type,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_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_ILBC_AUDIO_ENCODER_ILBC_CONFIG_H_
+#define API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_CONFIG_H_
+
+namespace webrtc {
+
+struct AudioEncoderIlbcConfig {
+  bool IsOk() const {
+    return (frame_size_ms == 20 || frame_size_ms == 30 || frame_size_ms == 40 ||
+            frame_size_ms == 60);
+  }
+  int frame_size_ms = 30;  // Valid values are 20, 30, 40, and 60 ms.
+  // Note that frame size 40 ms produces encodings with two 20 ms frames in
+  // them, and frame size 60 ms consists of two 30 ms frames.
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_CONFIG_H_