1 /* 2 * Copyright (c) 2012 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_ACM2_AUDIO_CODING_MODULE_IMPL_H_ 12 #define WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_ 13 14 #include <string> 15 #include <vector> 16 17 #include "webrtc/base/buffer.h" 18 #include "webrtc/base/scoped_ptr.h" 19 #include "webrtc/base/thread_annotations.h" 20 #include "webrtc/common_types.h" 21 #include "webrtc/engine_configurations.h" 22 #include "webrtc/modules/audio_coding/acm2/acm_receiver.h" 23 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h" 24 #include "webrtc/modules/audio_coding/acm2/codec_manager.h" 25 26 namespace webrtc { 27 28 class CriticalSectionWrapper; 29 class AudioCodingImpl; 30 31 namespace acm2 { 32 33 class AudioCodingModuleImpl final : public AudioCodingModule { 34 public: 35 friend webrtc::AudioCodingImpl; 36 37 explicit AudioCodingModuleImpl(const AudioCodingModule::Config& config); 38 ~AudioCodingModuleImpl() override; 39 40 ///////////////////////////////////////// 41 // Sender 42 // 43 44 // Can be called multiple times for Codec, CNG, RED. 45 int RegisterSendCodec(const CodecInst& send_codec) override; 46 47 void RegisterExternalSendCodec( 48 AudioEncoder* external_speech_encoder) override; 49 50 // Get current send codec. 51 rtc::Optional<CodecInst> SendCodec() const override; 52 53 // Get current send frequency. 54 int SendFrequency() const override; 55 56 // Sets the bitrate to the specified value in bits/sec. In case the codec does 57 // not support the requested value it will choose an appropriate value 58 // instead. 59 void SetBitRate(int bitrate_bps) override; 60 61 // Register a transport callback which will be 62 // called to deliver the encoded buffers. 63 int RegisterTransportCallback(AudioPacketizationCallback* transport) override; 64 65 // Add 10 ms of raw (PCM) audio data to the encoder. 66 int Add10MsData(const AudioFrame& audio_frame) override; 67 68 ///////////////////////////////////////// 69 // (RED) Redundant Coding 70 // 71 72 // Configure RED status i.e. on/off. 73 int SetREDStatus(bool enable_red) override; 74 75 // Get RED status. 76 bool REDStatus() const override; 77 78 ///////////////////////////////////////// 79 // (FEC) Forward Error Correction (codec internal) 80 // 81 82 // Configure FEC status i.e. on/off. 83 int SetCodecFEC(bool enabled_codec_fec) override; 84 85 // Get FEC status. 86 bool CodecFEC() const override; 87 88 // Set target packet loss rate 89 int SetPacketLossRate(int loss_rate) override; 90 91 ///////////////////////////////////////// 92 // (VAD) Voice Activity Detection 93 // and 94 // (CNG) Comfort Noise Generation 95 // 96 97 int SetVAD(bool enable_dtx = true, 98 bool enable_vad = false, 99 ACMVADMode mode = VADNormal) override; 100 101 int VAD(bool* dtx_enabled, 102 bool* vad_enabled, 103 ACMVADMode* mode) const override; 104 105 int RegisterVADCallback(ACMVADCallback* vad_callback) override; 106 107 ///////////////////////////////////////// 108 // Receiver 109 // 110 111 // Initialize receiver, resets codec database etc. 112 int InitializeReceiver() override; 113 114 // Get current receive frequency. 115 int ReceiveFrequency() const override; 116 117 // Get current playout frequency. 118 int PlayoutFrequency() const override; 119 120 // Register possible receive codecs, can be called multiple times, 121 // for codecs, CNG, DTMF, RED. 122 int RegisterReceiveCodec(const CodecInst& receive_codec) override; 123 124 int RegisterExternalReceiveCodec(int rtp_payload_type, 125 AudioDecoder* external_decoder, 126 int sample_rate_hz, 127 int num_channels, 128 const std::string& name) override; 129 130 // Get current received codec. 131 int ReceiveCodec(CodecInst* current_codec) const override; 132 133 // Incoming packet from network parsed and ready for decode. 134 int IncomingPacket(const uint8_t* incoming_payload, 135 const size_t payload_length, 136 const WebRtcRTPHeader& rtp_info) override; 137 138 // Incoming payloads, without rtp-info, the rtp-info will be created in ACM. 139 // One usage for this API is when pre-encoded files are pushed in ACM. 140 int IncomingPayload(const uint8_t* incoming_payload, 141 const size_t payload_length, 142 uint8_t payload_type, 143 uint32_t timestamp) override; 144 145 // Minimum playout delay. 146 int SetMinimumPlayoutDelay(int time_ms) override; 147 148 // Maximum playout delay. 149 int SetMaximumPlayoutDelay(int time_ms) override; 150 151 // Smallest latency NetEq will maintain. 152 int LeastRequiredDelayMs() const override; 153 154 // Get playout timestamp. 155 int PlayoutTimestamp(uint32_t* timestamp) override; 156 157 // Get 10 milliseconds of raw audio data to play out, and 158 // automatic resample to the requested frequency if > 0. 159 int PlayoutData10Ms(int desired_freq_hz, AudioFrame* audio_frame) override; 160 161 ///////////////////////////////////////// 162 // Statistics 163 // 164 165 int GetNetworkStatistics(NetworkStatistics* statistics) override; 166 167 int SetOpusApplication(OpusApplicationMode application) override; 168 169 // If current send codec is Opus, informs it about the maximum playback rate 170 // the receiver will render. 171 int SetOpusMaxPlaybackRate(int frequency_hz) override; 172 173 int EnableOpusDtx() override; 174 175 int DisableOpusDtx() override; 176 177 int UnregisterReceiveCodec(uint8_t payload_type) override; 178 179 int EnableNack(size_t max_nack_list_size) override; 180 181 void DisableNack() override; 182 183 std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const override; 184 185 void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const override; 186 187 private: 188 struct InputData { 189 uint32_t input_timestamp; 190 const int16_t* audio; 191 size_t length_per_channel; 192 size_t audio_channel; 193 // If a re-mix is required (up or down), this buffer will store a re-mixed 194 // version of the input. 195 int16_t buffer[WEBRTC_10MS_PCM_AUDIO]; 196 }; 197 198 // This member class writes values to the named UMA histogram, but only if 199 // the value has changed since the last time (and always for the first call). 200 class ChangeLogger { 201 public: 202 explicit ChangeLogger(const std::string& histogram_name) 203 : histogram_name_(histogram_name) {} 204 // Logs the new value if it is different from the last logged value, or if 205 // this is the first call. 206 void MaybeLog(int value); 207 208 private: 209 int last_value_ = 0; 210 int first_time_ = true; 211 const std::string histogram_name_; 212 }; 213 214 int Add10MsDataInternal(const AudioFrame& audio_frame, InputData* input_data) 215 EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_); 216 int Encode(const InputData& input_data) 217 EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_); 218 219 int InitializeReceiverSafe() EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_); 220 221 bool HaveValidEncoder(const char* caller_name) const 222 EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_); 223 224 // Preprocessing of input audio, including resampling and down-mixing if 225 // required, before pushing audio into encoder's buffer. 226 // 227 // in_frame: input audio-frame 228 // ptr_out: pointer to output audio_frame. If no preprocessing is required 229 // |ptr_out| will be pointing to |in_frame|, otherwise pointing to 230 // |preprocess_frame_|. 231 // 232 // Return value: 233 // -1: if encountering an error. 234 // 0: otherwise. 235 int PreprocessToAddData(const AudioFrame& in_frame, 236 const AudioFrame** ptr_out) 237 EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_); 238 239 // Change required states after starting to receive the codec corresponding 240 // to |index|. 241 int UpdateUponReceivingCodec(int index); 242 243 const rtc::scoped_ptr<CriticalSectionWrapper> acm_crit_sect_; 244 rtc::Buffer encode_buffer_ GUARDED_BY(acm_crit_sect_); 245 int id_; // TODO(henrik.lundin) Make const. 246 uint32_t expected_codec_ts_ GUARDED_BY(acm_crit_sect_); 247 uint32_t expected_in_ts_ GUARDED_BY(acm_crit_sect_); 248 ACMResampler resampler_ GUARDED_BY(acm_crit_sect_); 249 AcmReceiver receiver_; // AcmReceiver has it's own internal lock. 250 ChangeLogger bitrate_logger_ GUARDED_BY(acm_crit_sect_); 251 CodecManager codec_manager_ GUARDED_BY(acm_crit_sect_); 252 RentACodec rent_a_codec_ GUARDED_BY(acm_crit_sect_); 253 254 // This is to keep track of CN instances where we can send DTMFs. 255 uint8_t previous_pltype_ GUARDED_BY(acm_crit_sect_); 256 257 // Used when payloads are pushed into ACM without any RTP info 258 // One example is when pre-encoded bit-stream is pushed from 259 // a file. 260 // IMPORTANT: this variable is only used in IncomingPayload(), therefore, 261 // no lock acquired when interacting with this variable. If it is going to 262 // be used in other methods, locks need to be taken. 263 rtc::scoped_ptr<WebRtcRTPHeader> aux_rtp_header_; 264 265 bool receiver_initialized_ GUARDED_BY(acm_crit_sect_); 266 267 AudioFrame preprocess_frame_ GUARDED_BY(acm_crit_sect_); 268 bool first_10ms_data_ GUARDED_BY(acm_crit_sect_); 269 270 bool first_frame_ GUARDED_BY(acm_crit_sect_); 271 uint32_t last_timestamp_ GUARDED_BY(acm_crit_sect_); 272 uint32_t last_rtp_timestamp_ GUARDED_BY(acm_crit_sect_); 273 274 const rtc::scoped_ptr<CriticalSectionWrapper> callback_crit_sect_; 275 AudioPacketizationCallback* packetization_callback_ 276 GUARDED_BY(callback_crit_sect_); 277 ACMVADCallback* vad_callback_ GUARDED_BY(callback_crit_sect_); 278 }; 279 280 } // namespace acm2 281 } // namespace webrtc 282 283 #endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_ 284
