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Diffstat (limited to 'webrtc/modules/audio_coding/codecs/opus/include/audio_encoder_opus.h')
-rw-r--r-- | webrtc/modules/audio_coding/codecs/opus/include/audio_encoder_opus.h | 102 |
1 files changed, 0 insertions, 102 deletions
diff --git a/webrtc/modules/audio_coding/codecs/opus/include/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/include/audio_encoder_opus.h deleted file mode 100644 index 7f2b563fd9..0000000000 --- a/webrtc/modules/audio_coding/codecs/opus/include/audio_encoder_opus.h +++ /dev/null @@ -1,102 +0,0 @@ -/* - * Copyright (c) 2014 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 WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INCLUDE_AUDIO_ENCODER_OPUS_H_ -#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INCLUDE_AUDIO_ENCODER_OPUS_H_ - -#include <vector> - -#include "webrtc/base/constructormagic.h" -#include "webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h" -#include "webrtc/modules/audio_coding/codecs/audio_encoder.h" - -namespace webrtc { - -struct CodecInst; - -class AudioEncoderOpus final : public AudioEncoder { - public: - enum ApplicationMode { - kVoip = 0, - kAudio = 1, - }; - - struct Config { - bool IsOk() const; - int frame_size_ms = 20; - int num_channels = 1; - int payload_type = 120; - ApplicationMode application = kVoip; - int bitrate_bps = 64000; - bool fec_enabled = false; - int max_playback_rate_hz = 48000; - int complexity = kDefaultComplexity; - bool dtx_enabled = false; - - private: -#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM) - // If we are on Android, iOS and/or ARM, use a lower complexity setting as - // default, to save encoder complexity. - static const int kDefaultComplexity = 5; -#else - static const int kDefaultComplexity = 9; -#endif - }; - - explicit AudioEncoderOpus(const Config& config); - explicit AudioEncoderOpus(const CodecInst& codec_inst); - ~AudioEncoderOpus() override; - - size_t MaxEncodedBytes() const override; - int SampleRateHz() const override; - int NumChannels() const override; - size_t Num10MsFramesInNextPacket() const override; - size_t Max10MsFramesInAPacket() const override; - int GetTargetBitrate() const override; - - EncodedInfo EncodeInternal(uint32_t rtp_timestamp, - const int16_t* audio, - size_t max_encoded_bytes, - uint8_t* encoded) override; - - void Reset() override; - bool SetFec(bool enable) override; - - // Set Opus DTX. Once enabled, Opus stops transmission, when it detects voice - // being inactive. During that, it still sends 2 packets (one for content, one - // for signaling) about every 400 ms. - bool SetDtx(bool enable) override; - - bool SetApplication(Application application) override; - void SetMaxPlaybackRate(int frequency_hz) override; - void SetProjectedPacketLossRate(double fraction) override; - void SetTargetBitrate(int target_bps) override; - - // Getters for testing. - double packet_loss_rate() const { return packet_loss_rate_; } - ApplicationMode application() const { return config_.application; } - bool dtx_enabled() const { return config_.dtx_enabled; } - - private: - int Num10msFramesPerPacket() const; - int SamplesPer10msFrame() const; - bool RecreateEncoderInstance(const Config& config); - - Config config_; - double packet_loss_rate_; - std::vector<int16_t> input_buffer_; - OpusEncInst* inst_; - uint32_t first_timestamp_in_buffer_; - RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus); -}; - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INCLUDE_AUDIO_ENCODER_OPUS_H_ |