Repo created

TMessagesProj/jni/voip/webrtc/api/audio_codecs/L16/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_L16") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_encoder_L16.cc",
+    "audio_encoder_L16.h",
+  ]
+  deps = [
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:pcm16b",
+    "../../../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_L16") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_decoder_L16.cc",
+    "audio_decoder_L16.h",
+  ]
+  deps = [
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:pcm16b",
+    "../../../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/L16/audio_decoder_L16.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/L16/audio_decoder_L16.h"
+
+#include <memory>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
+#include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+absl::optional<AudioDecoderL16::Config> AudioDecoderL16::SdpToConfig(
+    const SdpAudioFormat& format) {
+  Config config;
+  config.sample_rate_hz = format.clockrate_hz;
+  config.num_channels = rtc::checked_cast<int>(format.num_channels);
+  if (absl::EqualsIgnoreCase(format.name, "L16") && config.IsOk()) {
+    return config;
+  }
+  return absl::nullopt;
+}
+
+void AudioDecoderL16::AppendSupportedDecoders(
+    std::vector<AudioCodecSpec>* specs) {
+  Pcm16BAppendSupportedCodecSpecs(specs);
+}
+
+std::unique_ptr<AudioDecoder> AudioDecoderL16::MakeAudioDecoder(
+    const Config& config,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  if (!config.IsOk()) {
+    return nullptr;
+  }
+  return std::make_unique<AudioDecoderPcm16B>(config.sample_rate_hz,
+                                              config.num_channels);
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/L16/audio_decoder_L16.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_L16_AUDIO_DECODER_L16_H_
+#define API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_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 {
+
+// L16 decoder API for use as a template parameter to
+// CreateAudioDecoderFactory<...>().
+struct RTC_EXPORT AudioDecoderL16 {
+  struct Config {
+    bool IsOk() const {
+      return (sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
+              sample_rate_hz == 32000 || sample_rate_hz == 48000) &&
+             (num_channels >= 1 &&
+              num_channels <= AudioDecoder::kMaxNumberOfChannels);
+    }
+    int sample_rate_hz = 8000;
+    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(
+      const Config& config,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/L16/audio_encoder_L16.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/L16/audio_encoder_L16.h"
+
+#include <memory>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
+#include "modules/audio_coding/codecs/pcm16b/pcm16b_common.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<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig(
+    const SdpAudioFormat& format) {
+  if (!rtc::IsValueInRangeForNumericType<int>(format.num_channels)) {
+    RTC_DCHECK_NOTREACHED();
+    return absl::nullopt;
+  }
+  Config config;
+  config.sample_rate_hz = format.clockrate_hz;
+  config.num_channels = rtc::dchecked_cast<int>(format.num_channels);
+  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 (absl::EqualsIgnoreCase(format.name, "L16") && config.IsOk()) {
+    return config;
+  }
+  return absl::nullopt;
+}
+
+void AudioEncoderL16::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  Pcm16BAppendSupportedCodecSpecs(specs);
+}
+
+AudioCodecInfo AudioEncoderL16::QueryAudioEncoder(
+    const AudioEncoderL16::Config& config) {
+  RTC_DCHECK(config.IsOk());
+  return {config.sample_rate_hz,
+          rtc::dchecked_cast<size_t>(config.num_channels),
+          config.sample_rate_hz * config.num_channels * 16};
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderL16::MakeAudioEncoder(
+    const AudioEncoderL16::Config& config,
+    int payload_type,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  AudioEncoderPcm16B::Config c;
+  c.sample_rate_hz = config.sample_rate_hz;
+  c.num_channels = config.num_channels;
+  c.frame_size_ms = config.frame_size_ms;
+  c.payload_type = payload_type;
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  return std::make_unique<AudioEncoderPcm16B>(c);
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/L16/audio_encoder_L16.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_L16_AUDIO_ENCODER_L16_H_
+#define API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_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 {
+
+// L16 encoder API for use as a template parameter to
+// CreateAudioEncoderFactory<...>().
+struct RTC_EXPORT AudioEncoderL16 {
+  struct Config {
+    bool IsOk() const {
+      return (sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
+              sample_rate_hz == 32000 || sample_rate_hz == 48000) &&
+             num_channels >= 1 &&
+             num_channels <= AudioEncoder::kMaxNumberOfChannels &&
+             frame_size_ms > 0 && frame_size_ms <= 120 &&
+             frame_size_ms % 10 == 0;
+    }
+    int sample_rate_hz = 8000;
+    int num_channels = 1;
+    int frame_size_ms = 10;
+  };
+  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_L16_AUDIO_ENCODER_L16_H_