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 WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_ 12 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_ 13 14 #include <vector> 15 16 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" 17 #include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h" 18 19 namespace webrtc { 20 21 struct CodecInst; 22 23 template <typename T> 24 class AudioEncoderIsacT final : public AudioEncoder { 25 public: 26 // Allowed combinations of sample rate, frame size, and bit rate are 27 // - 16000 Hz, 30 ms, 10000-32000 bps 28 // - 16000 Hz, 60 ms, 10000-32000 bps 29 // - 32000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support) 30 struct Config { 31 bool IsOk() const; 32 33 LockedIsacBandwidthInfo* bwinfo = nullptr; 34 35 int payload_type = 103; 36 int sample_rate_hz = 16000; 37 int frame_size_ms = 30; 38 int bit_rate = kDefaultBitRate; // Limit on the short-term average bit 39 // rate, in bits/s. 40 int max_payload_size_bytes = -1; 41 int max_bit_rate = -1; 42 43 // If true, the encoder will dynamically adjust frame size and bit rate; 44 // the configured values are then merely the starting point. 45 bool adaptive_mode = false; 46 47 // In adaptive mode, prevent adaptive changes to the frame size. (Not used 48 // in nonadaptive mode.) 49 bool enforce_frame_size = false; 50 }; 51 52 explicit AudioEncoderIsacT(const Config& config); 53 explicit AudioEncoderIsacT(const CodecInst& codec_inst, 54 LockedIsacBandwidthInfo* bwinfo); 55 ~AudioEncoderIsacT() override; 56 57 size_t MaxEncodedBytes() const override; 58 int SampleRateHz() const override; 59 size_t NumChannels() const override; 60 size_t Num10MsFramesInNextPacket() const override; 61 size_t Max10MsFramesInAPacket() const override; 62 int GetTargetBitrate() const override; 63 EncodedInfo EncodeInternal(uint32_t rtp_timestamp, 64 rtc::ArrayView<const int16_t> audio, 65 size_t max_encoded_bytes, 66 uint8_t* encoded) override; 67 void Reset() override; 68 69 private: 70 // This value is taken from STREAM_SIZE_MAX_60 for iSAC float (60 ms) and 71 // STREAM_MAXW16_60MS for iSAC fix (60 ms). 72 static const size_t kSufficientEncodeBufferSizeBytes = 400; 73 74 static const int kDefaultBitRate = 32000; 75 76 // Recreate the iSAC encoder instance with the given settings, and save them. 77 void RecreateEncoderInstance(const Config& config); 78 79 Config config_; 80 typename T::instance_type* isac_state_ = nullptr; 81 LockedIsacBandwidthInfo* bwinfo_ = nullptr; 82 83 // Have we accepted input but not yet emitted it in a packet? 84 bool packet_in_progress_ = false; 85 86 // Timestamp of the first input of the currently in-progress packet. 87 uint32_t packet_timestamp_; 88 89 // Timestamp of the previously encoded packet. 90 uint32_t last_encoded_timestamp_; 91 92 RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIsacT); 93 }; 94 95 } // namespace webrtc 96 97 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_ 98
