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_OPUS_AUDIO_ENCODER_OPUS_H_ 12 #define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_ 13 14 #include <functional> 15 #include <memory> 16 #include <string> 17 #include <vector> 18 19 #include "absl/types/optional.h" 20 #include "api/audio_codecs/audio_encoder.h" 21 #include "api/audio_codecs/audio_format.h" 22 #include "api/audio_codecs/opus/audio_encoder_opus_config.h" 23 #include "common_audio/smoothing_filter.h" 24 #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h" 25 #include "modules/audio_coding/codecs/opus/opus_interface.h" 26 #include "rtc_base/constructor_magic.h" 27 28 namespace webrtc { 29 30 class RtcEventLog; 31 32 class AudioEncoderOpusImpl final : public AudioEncoder { 33 public: 34 // Returns empty if the current bitrate falls within the hysteresis window, 35 // defined by complexity_threshold_bps +/- complexity_threshold_window_bps. 36 // Otherwise, returns the current complexity depending on whether the 37 // current bitrate is above or below complexity_threshold_bps. 38 static absl::optional<int> GetNewComplexity( 39 const AudioEncoderOpusConfig& config); 40 41 // Returns OPUS_AUTO if the the current bitrate is above wideband threshold. 42 // Returns empty if it is below, but bandwidth coincides with the desired one. 43 // Otherwise returns the desired bandwidth. 44 static absl::optional<int> GetNewBandwidth( 45 const AudioEncoderOpusConfig& config, 46 OpusEncInst* inst); 47 48 using AudioNetworkAdaptorCreator = 49 std::function<std::unique_ptr<AudioNetworkAdaptor>(const std::string&, 50 RtcEventLog*)>; 51 52 AudioEncoderOpusImpl(const AudioEncoderOpusConfig& config, int payload_type); 53 54 // Dependency injection for testing. 55 AudioEncoderOpusImpl( 56 const AudioEncoderOpusConfig& config, 57 int payload_type, 58 const AudioNetworkAdaptorCreator& audio_network_adaptor_creator, 59 std::unique_ptr<SmoothingFilter> bitrate_smoother); 60 61 AudioEncoderOpusImpl(int payload_type, const SdpAudioFormat& format); 62 ~AudioEncoderOpusImpl() override; 63 64 int SampleRateHz() const override; 65 size_t NumChannels() const override; 66 int RtpTimestampRateHz() const override; 67 size_t Num10MsFramesInNextPacket() const override; 68 size_t Max10MsFramesInAPacket() const override; 69 int GetTargetBitrate() const override; 70 71 void Reset() override; 72 bool SetFec(bool enable) override; 73 74 // Set Opus DTX. Once enabled, Opus stops transmission, when it detects 75 // voice being inactive. During that, it still sends 2 packets (one for 76 // content, one for signaling) about every 400 ms. 77 bool SetDtx(bool enable) override; 78 bool GetDtx() const override; 79 80 bool SetApplication(Application application) override; 81 void SetMaxPlaybackRate(int frequency_hz) override; 82 bool EnableAudioNetworkAdaptor(const std::string& config_string, 83 RtcEventLog* event_log) override; 84 void DisableAudioNetworkAdaptor() override; 85 void OnReceivedUplinkPacketLossFraction( 86 float uplink_packet_loss_fraction) override; 87 void OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) override; 88 void OnReceivedUplinkBandwidth( 89 int target_audio_bitrate_bps, 90 absl::optional<int64_t> bwe_period_ms) override; 91 void OnReceivedUplinkAllocation(BitrateAllocationUpdate update) override; 92 void OnReceivedRtt(int rtt_ms) override; 93 void OnReceivedOverhead(size_t overhead_bytes_per_packet) override; 94 void SetReceiverFrameLengthRange(int min_frame_length_ms, 95 int max_frame_length_ms) override; 96 ANAStats GetANAStats() const override; 97 absl::optional<std::pair<TimeDelta, TimeDelta> > GetFrameLengthRange() 98 const override; supported_frame_lengths_ms()99 rtc::ArrayView<const int> supported_frame_lengths_ms() const { 100 return config_.supported_frame_lengths_ms; 101 } 102 103 // Getters for testing. packet_loss_rate()104 float packet_loss_rate() const { return packet_loss_rate_; } application()105 AudioEncoderOpusConfig::ApplicationMode application() const { 106 return config_.application; 107 } fec_enabled()108 bool fec_enabled() const { return config_.fec_enabled; } num_channels_to_encode()109 size_t num_channels_to_encode() const { return num_channels_to_encode_; } next_frame_length_ms()110 int next_frame_length_ms() const { return next_frame_length_ms_; } 111 112 protected: 113 EncodedInfo EncodeImpl(uint32_t rtp_timestamp, 114 rtc::ArrayView<const int16_t> audio, 115 rtc::Buffer* encoded) override; 116 117 private: 118 class PacketLossFractionSmoother; 119 120 static absl::optional<AudioEncoderOpusConfig> SdpToConfig( 121 const SdpAudioFormat& format); 122 static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs); 123 static AudioCodecInfo QueryAudioEncoder(const AudioEncoderOpusConfig& config); 124 static std::unique_ptr<AudioEncoder> MakeAudioEncoder( 125 const AudioEncoderOpusConfig&, 126 int payload_type); 127 128 size_t Num10msFramesPerPacket() const; 129 size_t SamplesPer10msFrame() const; 130 size_t SufficientOutputBufferSize() const; 131 bool RecreateEncoderInstance(const AudioEncoderOpusConfig& config); 132 void SetFrameLength(int frame_length_ms); 133 void SetNumChannelsToEncode(size_t num_channels_to_encode); 134 void SetProjectedPacketLossRate(float fraction); 135 136 void OnReceivedUplinkBandwidth( 137 int target_audio_bitrate_bps, 138 absl::optional<int64_t> bwe_period_ms, 139 absl::optional<int64_t> link_capacity_allocation); 140 141 // TODO(minyue): remove "override" when we can deprecate 142 // |AudioEncoder::SetTargetBitrate|. 143 void SetTargetBitrate(int target_bps) override; 144 145 void ApplyAudioNetworkAdaptor(); 146 std::unique_ptr<AudioNetworkAdaptor> DefaultAudioNetworkAdaptorCreator( 147 const std::string& config_string, 148 RtcEventLog* event_log) const; 149 150 void MaybeUpdateUplinkBandwidth(); 151 152 AudioEncoderOpusConfig config_; 153 const int payload_type_; 154 const bool send_side_bwe_with_overhead_; 155 const bool use_stable_target_for_adaptation_; 156 const bool adjust_bandwidth_; 157 bool bitrate_changed_; 158 // A multiplier for bitrates at 5 kbps and higher. The target bitrate 159 // will be multiplied by these multipliers, each multiplier is applied to a 160 // 1 kbps range. 161 std::vector<float> bitrate_multipliers_; 162 float packet_loss_rate_; 163 std::vector<int16_t> input_buffer_; 164 OpusEncInst* inst_; 165 uint32_t first_timestamp_in_buffer_; 166 size_t num_channels_to_encode_; 167 int next_frame_length_ms_; 168 int complexity_; 169 std::unique_ptr<PacketLossFractionSmoother> packet_loss_fraction_smoother_; 170 const AudioNetworkAdaptorCreator audio_network_adaptor_creator_; 171 std::unique_ptr<AudioNetworkAdaptor> audio_network_adaptor_; 172 absl::optional<size_t> overhead_bytes_per_packet_; 173 const std::unique_ptr<SmoothingFilter> bitrate_smoother_; 174 absl::optional<int64_t> bitrate_smoother_last_update_time_; 175 176 friend struct AudioEncoderOpus; 177 RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpusImpl); 178 }; 179 180 } // namespace webrtc 181 182 #endif // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_ 183