Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
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modules/audio_coding/codecs/cng/audio_encoder_cng.h
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modules/audio_coding/codecs/cng/audio_encoder_cng.h
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/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
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#define WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
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#include <memory>
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#include <vector>
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#include "webrtc/api/audio_codecs/audio_encoder.h"
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#include "webrtc/common_audio/vad/include/vad.h"
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#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
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#include "webrtc/rtc_base/constructormagic.h"
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namespace webrtc {
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class Vad;
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class AudioEncoderCng final : public AudioEncoder {
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public:
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struct Config {
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Config();
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Config(Config&&);
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~Config();
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bool IsOk() const;
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size_t num_channels = 1;
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int payload_type = 13;
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std::unique_ptr<AudioEncoder> speech_encoder;
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Vad::Aggressiveness vad_mode = Vad::kVadNormal;
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int sid_frame_interval_ms = 100;
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int num_cng_coefficients = 8;
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// The Vad pointer is mainly for testing. If a NULL pointer is passed, the
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// AudioEncoderCng creates (and destroys) a Vad object internally. If an
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// object is passed, the AudioEncoderCng assumes ownership of the Vad
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// object.
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Vad* vad = nullptr;
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};
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explicit AudioEncoderCng(Config&& config);
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~AudioEncoderCng() override;
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int SampleRateHz() const override;
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size_t NumChannels() const override;
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int RtpTimestampRateHz() const override;
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size_t Num10MsFramesInNextPacket() const override;
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size_t Max10MsFramesInAPacket() const override;
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int GetTargetBitrate() const override;
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EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
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rtc::ArrayView<const int16_t> audio,
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rtc::Buffer* encoded) override;
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void Reset() override;
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bool SetFec(bool enable) override;
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bool SetDtx(bool enable) override;
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bool SetApplication(Application application) override;
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void SetMaxPlaybackRate(int frequency_hz) override;
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rtc::ArrayView<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders()
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override;
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void OnReceivedUplinkPacketLossFraction(
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float uplink_packet_loss_fraction) override;
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void OnReceivedUplinkRecoverablePacketLossFraction(
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float uplink_recoverable_packet_loss_fraction) override;
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void OnReceivedUplinkBandwidth(
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int target_audio_bitrate_bps,
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rtc::Optional<int64_t> bwe_period_ms) override;
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private:
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EncodedInfo EncodePassive(size_t frames_to_encode,
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rtc::Buffer* encoded);
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EncodedInfo EncodeActive(size_t frames_to_encode,
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rtc::Buffer* encoded);
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size_t SamplesPer10msFrame() const;
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std::unique_ptr<AudioEncoder> speech_encoder_;
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const int cng_payload_type_;
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const int num_cng_coefficients_;
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const int sid_frame_interval_ms_;
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std::vector<int16_t> speech_buffer_;
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std::vector<uint32_t> rtp_timestamps_;
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bool last_frame_active_;
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std::unique_ptr<Vad> vad_;
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std::unique_ptr<ComfortNoiseEncoder> cng_encoder_;
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RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderCng);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
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