1 /* 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_ 12 #define MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_ 13 14 #include <memory> 15 #include <utility> 16 17 #include "absl/types/optional.h" 18 #include "api/audio_codecs/audio_encoder.h" 19 #include "api/audio_codecs/g722/audio_encoder_g722_config.h" 20 #include "api/units/time_delta.h" 21 #include "modules/audio_coding/codecs/g722/g722_interface.h" 22 #include "rtc_base/buffer.h" 23 24 namespace webrtc { 25 26 class AudioEncoderG722Impl final : public AudioEncoder { 27 public: 28 AudioEncoderG722Impl(const AudioEncoderG722Config& config, int payload_type); 29 ~AudioEncoderG722Impl() override; 30 31 AudioEncoderG722Impl(const AudioEncoderG722Impl&) = delete; 32 AudioEncoderG722Impl& operator=(const AudioEncoderG722Impl&) = delete; 33 34 int SampleRateHz() const override; 35 size_t NumChannels() const override; 36 int RtpTimestampRateHz() const override; 37 size_t Num10MsFramesInNextPacket() const override; 38 size_t Max10MsFramesInAPacket() const override; 39 int GetTargetBitrate() const override; 40 void Reset() override; 41 absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange() 42 const override; 43 44 protected: 45 EncodedInfo EncodeImpl(uint32_t rtp_timestamp, 46 rtc::ArrayView<const int16_t> audio, 47 rtc::Buffer* encoded) override; 48 49 private: 50 // The encoder state for one channel. 51 struct EncoderState { 52 G722EncInst* encoder; 53 std::unique_ptr<int16_t[]> speech_buffer; // Queued up for encoding. 54 rtc::Buffer encoded_buffer; // Already encoded. 55 EncoderState(); 56 ~EncoderState(); 57 }; 58 59 size_t SamplesPerChannel() const; 60 61 const size_t num_channels_; 62 const int payload_type_; 63 const size_t num_10ms_frames_per_packet_; 64 size_t num_10ms_frames_buffered_; 65 uint32_t first_timestamp_in_buffer_; 66 const std::unique_ptr<EncoderState[]> encoders_; 67 rtc::Buffer interleave_buffer_; 68 }; 69 70 } // namespace webrtc 71 #endif // MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_ 72