| /* |
| * Copyright (c) 2012 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_INCLUDE_AUDIO_CODING_MODULE_H_ |
| #define WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_ |
| |
| #include <memory> |
| #include <string> |
| #include <vector> |
| |
| #include "webrtc/api/audio_codecs/audio_decoder_factory.h" |
| #include "webrtc/api/audio_codecs/audio_encoder.h" |
| #include "webrtc/api/optional.h" |
| #include "webrtc/common_types.h" |
| #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" |
| #include "webrtc/modules/audio_coding/neteq/include/neteq.h" |
| #include "webrtc/modules/include/module.h" |
| #include "webrtc/rtc_base/deprecation.h" |
| #include "webrtc/rtc_base/function_view.h" |
| #include "webrtc/system_wrappers/include/clock.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| |
| // forward declarations |
| struct CodecInst; |
| struct WebRtcRTPHeader; |
| class AudioDecoder; |
| class AudioEncoder; |
| class AudioFrame; |
| class RTPFragmentationHeader; |
| |
| #define WEBRTC_10MS_PCM_AUDIO 960 // 16 bits super wideband 48 kHz |
| |
| // Callback class used for sending data ready to be packetized |
| class AudioPacketizationCallback { |
| public: |
| virtual ~AudioPacketizationCallback() {} |
| |
| virtual int32_t SendData(FrameType frame_type, |
| uint8_t payload_type, |
| uint32_t timestamp, |
| const uint8_t* payload_data, |
| size_t payload_len_bytes, |
| const RTPFragmentationHeader* fragmentation) = 0; |
| }; |
| |
| // Callback class used for reporting VAD decision |
| class ACMVADCallback { |
| public: |
| virtual ~ACMVADCallback() {} |
| |
| virtual int32_t InFrameType(FrameType frame_type) = 0; |
| }; |
| |
| class AudioCodingModule { |
| protected: |
| AudioCodingModule() {} |
| |
| public: |
| struct Config { |
| Config(); |
| Config(const Config&); |
| ~Config(); |
| |
| int id; |
| NetEq::Config neteq_config; |
| Clock* clock; |
| rtc::scoped_refptr<AudioDecoderFactory> decoder_factory; |
| }; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // Creation and destruction of a ACM. |
| // |
| // The second method is used for testing where a simulated clock can be |
| // injected into ACM. ACM will take the ownership of the object clock and |
| // delete it when destroyed. |
| // |
| static AudioCodingModule* Create(int id); |
| static AudioCodingModule* Create(int id, Clock* clock); |
| static AudioCodingModule* Create(const Config& config); |
| virtual ~AudioCodingModule() = default; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // Utility functions |
| // |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // uint8_t NumberOfCodecs() |
| // Returns number of supported codecs. |
| // |
| // Return value: |
| // number of supported codecs. |
| /// |
| static int NumberOfCodecs(); |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int32_t Codec() |
| // Get supported codec with list number. |
| // |
| // Input: |
| // -list_id : list number. |
| // |
| // Output: |
| // -codec : a structure where the parameters of the codec, |
| // given by list number is written to. |
| // |
| // Return value: |
| // -1 if the list number (list_id) is invalid. |
| // 0 if succeeded. |
| // |
| static int Codec(int list_id, CodecInst* codec); |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int32_t Codec() |
| // Get supported codec with the given codec name, sampling frequency, and |
| // a given number of channels. |
| // |
| // Input: |
| // -payload_name : name of the codec. |
| // -sampling_freq_hz : sampling frequency of the codec. Note! for RED |
| // a sampling frequency of -1 is a valid input. |
| // -channels : number of channels ( 1 - mono, 2 - stereo). |
| // |
| // Output: |
| // -codec : a structure where the function returns the |
| // default parameters of the codec. |
| // |
| // Return value: |
| // -1 if no codec matches the given parameters. |
| // 0 if succeeded. |
| // |
| static int Codec(const char* payload_name, CodecInst* codec, |
| int sampling_freq_hz, size_t channels); |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int32_t Codec() |
| // |
| // Returns the list number of the given codec name, sampling frequency, and |
| // a given number of channels. |
| // |
| // Input: |
| // -payload_name : name of the codec. |
| // -sampling_freq_hz : sampling frequency of the codec. Note! for RED |
| // a sampling frequency of -1 is a valid input. |
| // -channels : number of channels ( 1 - mono, 2 - stereo). |
| // |
| // Return value: |
| // if the codec is found, the index of the codec in the list, |
| // -1 if the codec is not found. |
| // |
| static int Codec(const char* payload_name, int sampling_freq_hz, |
| size_t channels); |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // bool IsCodecValid() |
| // Checks the validity of the parameters of the given codec. |
| // |
| // Input: |
| // -codec : the structure which keeps the parameters of the |
| // codec. |
| // |
| // Return value: |
| // true if the parameters are valid, |
| // false if any parameter is not valid. |
| // |
| static bool IsCodecValid(const CodecInst& codec); |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // Sender |
| // |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int32_t RegisterSendCodec() |
| // Registers a codec, specified by |send_codec|, as sending codec. |
| // This API can be called multiple of times to register Codec. The last codec |
| // registered overwrites the previous ones. |
| // The API can also be used to change payload type for CNG and RED, which are |
| // registered by default to default payload types. |
| // Note that registering CNG and RED won't overwrite speech codecs. |
| // This API can be called to set/change the send payload-type, frame-size |
| // or encoding rate (if applicable for the codec). |
| // |
| // Note: If a stereo codec is registered as send codec, VAD/DTX will |
| // automatically be turned off, since it is not supported for stereo sending. |
| // |
| // Note: If a secondary encoder is already registered, and the new send-codec |
| // has a sampling rate that does not match the secondary encoder, the |
| // secondary encoder will be unregistered. |
| // |
| // Input: |
| // -send_codec : Parameters of the codec to be registered, c.f. |
| // common_types.h for the definition of |
| // CodecInst. |
| // |
| // Return value: |
| // -1 if failed to initialize, |
| // 0 if succeeded. |
| // |
| virtual int32_t RegisterSendCodec(const CodecInst& send_codec) = 0; |
| |
| // Registers |external_speech_encoder| as encoder. The new encoder will |
| // replace any previously registered speech encoder (internal or external). |
| virtual void RegisterExternalSendCodec( |
| AudioEncoder* external_speech_encoder) = 0; |
| |
| // |modifier| is called exactly once with one argument: a pointer to the |
| // unique_ptr that holds the current encoder (which is null if there is no |
| // current encoder). For the duration of the call, |modifier| has exclusive |
| // access to the unique_ptr; it may call the encoder, steal the encoder and |
| // replace it with another encoder or with nullptr, etc. |
| virtual void ModifyEncoder( |
| rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) = 0; |
| |
| // |modifier| is called exactly once with one argument: a const pointer to the |
| // current encoder (which is null if there is no current encoder). |
| virtual void QueryEncoder( |
| rtc::FunctionView<void(AudioEncoder const*)> query) = 0; |
| |
| // Utility method for simply replacing the existing encoder with a new one. |
| void SetEncoder(std::unique_ptr<AudioEncoder> new_encoder) { |
| ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { |
| *encoder = std::move(new_encoder); |
| }); |
| } |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int32_t SendCodec() |
| // Get parameters for the codec currently registered as send codec. |
| // |
| // Return value: |
| // The send codec, or nothing if we don't have one |
| // |
| virtual rtc::Optional<CodecInst> SendCodec() const = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int32_t SendFrequency() |
| // Get the sampling frequency of the current encoder in Hertz. |
| // |
| // Return value: |
| // positive; sampling frequency [Hz] of the current encoder. |
| // -1 if an error has happened. |
| // |
| virtual int32_t SendFrequency() const = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // Sets the bitrate to the specified value in bits/sec. If the value is not |
| // supported by the codec, it will choose another appropriate value. |
| // |
| // This is only used in test code that rely on old ACM APIs. |
| // TODO(minyue): Remove it when possible. |
| virtual void SetBitRate(int bitrate_bps) = 0; |
| |
| // int32_t RegisterTransportCallback() |
| // Register a transport callback which will be called to deliver |
| // the encoded buffers whenever Process() is called and a |
| // bit-stream is ready. |
| // |
| // Input: |
| // -transport : pointer to the callback class |
| // transport->SendData() is called whenever |
| // Process() is called and bit-stream is ready |
| // to deliver. |
| // |
| // Return value: |
| // -1 if the transport callback could not be registered |
| // 0 if registration is successful. |
| // |
| virtual int32_t RegisterTransportCallback( |
| AudioPacketizationCallback* transport) = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int32_t Add10MsData() |
| // Add 10MS of raw (PCM) audio data and encode it. If the sampling |
| // frequency of the audio does not match the sampling frequency of the |
| // current encoder ACM will resample the audio. If an encoded packet was |
| // produced, it will be delivered via the callback object registered using |
| // RegisterTransportCallback, and the return value from this function will |
| // be the number of bytes encoded. |
| // |
| // Input: |
| // -audio_frame : the input audio frame, containing raw audio |
| // sampling frequency etc., |
| // c.f. module_common_types.h for definition of |
| // AudioFrame. |
| // |
| // Return value: |
| // >= 0 number of bytes encoded. |
| // -1 some error occurred. |
| // |
| virtual int32_t Add10MsData(const AudioFrame& audio_frame) = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // (RED) Redundant Coding |
| // |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int32_t SetREDStatus() |
| // configure RED status i.e. on/off. |
| // |
| // RFC 2198 describes a solution which has a single payload type which |
| // signifies a packet with redundancy. That packet then becomes a container, |
| // encapsulating multiple payloads into a single RTP packet. |
| // Such a scheme is flexible, since any amount of redundancy may be |
| // encapsulated within a single packet. There is, however, a small overhead |
| // since each encapsulated payload must be preceded by a header indicating |
| // the type of data enclosed. |
| // |
| // Input: |
| // -enable_red : if true RED is enabled, otherwise RED is |
| // disabled. |
| // |
| // Return value: |
| // -1 if failed to set RED status, |
| // 0 if succeeded. |
| // |
| virtual int32_t SetREDStatus(bool enable_red) = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // bool REDStatus() |
| // Get RED status |
| // |
| // Return value: |
| // true if RED is enabled, |
| // false if RED is disabled. |
| // |
| virtual bool REDStatus() const = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // (FEC) Forward Error Correction (codec internal) |
| // |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int32_t SetCodecFEC() |
| // Configures codec internal FEC status i.e. on/off. No effects on codecs that |
| // do not provide internal FEC. |
| // |
| // Input: |
| // -enable_fec : if true FEC will be enabled otherwise the FEC is |
| // disabled. |
| // |
| // Return value: |
| // -1 if failed, or the codec does not support FEC |
| // 0 if succeeded. |
| // |
| virtual int SetCodecFEC(bool enable_codec_fec) = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // bool CodecFEC() |
| // Gets status of codec internal FEC. |
| // |
| // Return value: |
| // true if FEC is enabled, |
| // false if FEC is disabled. |
| // |
| virtual bool CodecFEC() const = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int SetPacketLossRate() |
| // Sets expected packet loss rate for encoding. Some encoders provide packet |
| // loss gnostic encoding to make stream less sensitive to packet losses, |
| // through e.g., FEC. No effects on codecs that do not provide such encoding. |
| // |
| // Input: |
| // -packet_loss_rate : expected packet loss rate (0 -- 100 inclusive). |
| // |
| // Return value |
| // -1 if failed to set packet loss rate, |
| // 0 if succeeded. |
| // |
| // This is only used in test code that rely on old ACM APIs. |
| // TODO(minyue): Remove it when possible. |
| virtual int SetPacketLossRate(int packet_loss_rate) = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // (VAD) Voice Activity Detection |
| // |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int32_t SetVAD() |
| // If DTX is enabled & the codec does not have internal DTX/VAD |
| // WebRtc VAD will be automatically enabled and |enable_vad| is ignored. |
| // |
| // If DTX is disabled but VAD is enabled no DTX packets are send, |
| // regardless of whether the codec has internal DTX/VAD or not. In this |
| // case, WebRtc VAD is running to label frames as active/in-active. |
| // |
| // NOTE! VAD/DTX is not supported when sending stereo. |
| // |
| // Inputs: |
| // -enable_dtx : if true DTX is enabled, |
| // otherwise DTX is disabled. |
| // -enable_vad : if true VAD is enabled, |
| // otherwise VAD is disabled. |
| // -vad_mode : determines the aggressiveness of VAD. A more |
| // aggressive mode results in more frames labeled |
| // as in-active, c.f. definition of |
| // ACMVADMode in audio_coding_module_typedefs.h |
| // for valid values. |
| // |
| // Return value: |
| // -1 if failed to set up VAD/DTX, |
| // 0 if succeeded. |
| // |
| virtual int32_t SetVAD(const bool enable_dtx = true, |
| const bool enable_vad = false, |
| const ACMVADMode vad_mode = VADNormal) = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int32_t VAD() |
| // Get VAD status. |
| // |
| // Outputs: |
| // -dtx_enabled : is set to true if DTX is enabled, otherwise |
| // is set to false. |
| // -vad_enabled : is set to true if VAD is enabled, otherwise |
| // is set to false. |
| // -vad_mode : is set to the current aggressiveness of VAD. |
| // |
| // Return value: |
| // -1 if fails to retrieve the setting of DTX/VAD, |
| // 0 if succeeded. |
| // |
| virtual int32_t VAD(bool* dtx_enabled, bool* vad_enabled, |
| ACMVADMode* vad_mode) const = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int32_t RegisterVADCallback() |
| // Call this method to register a callback function which is called |
| // any time that ACM encounters an empty frame. That is a frame which is |
| // recognized inactive. Depending on the codec WebRtc VAD or internal codec |
| // VAD is employed to identify a frame as active/inactive. |
| // |
| // Input: |
| // -vad_callback : pointer to a callback function. |
| // |
| // Return value: |
| // -1 if failed to register the callback function. |
| // 0 if the callback function is registered successfully. |
| // |
| virtual int32_t RegisterVADCallback(ACMVADCallback* vad_callback) = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // Receiver |
| // |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int32_t InitializeReceiver() |
| // Any decoder-related state of ACM will be initialized to the |
| // same state when ACM is created. This will not interrupt or |
| // effect encoding functionality of ACM. ACM would lose all the |
| // decoding-related settings by calling this function. |
| // For instance, all registered codecs are deleted and have to be |
| // registered again. |
| // |
| // Return value: |
| // -1 if failed to initialize, |
| // 0 if succeeded. |
| // |
| virtual int32_t InitializeReceiver() = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int32_t ReceiveFrequency() |
| // Get sampling frequency of the last received payload. |
| // |
| // Return value: |
| // non-negative the sampling frequency in Hertz. |
| // -1 if an error has occurred. |
| // |
| virtual int32_t ReceiveFrequency() const = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int32_t PlayoutFrequency() |
| // Get sampling frequency of audio played out. |
| // |
| // Return value: |
| // the sampling frequency in Hertz. |
| // |
| virtual int32_t PlayoutFrequency() const = 0; |
| |
| // Replace any existing decoders with the given payload type -> decoder map. |
| virtual void SetReceiveCodecs( |
| const std::map<int, SdpAudioFormat>& codecs) = 0; |
| |
| // Registers a decoder for the given payload type. Returns true iff |
| // successful. |
| virtual bool RegisterReceiveCodec(int rtp_payload_type, |
| const SdpAudioFormat& audio_format) = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int32_t RegisterReceiveCodec() |
| // Register possible decoders, can be called multiple times for |
| // codecs, CNG-NB, CNG-WB, CNG-SWB, AVT and RED. |
| // |
| // Input: |
| // -receive_codec : parameters of the codec to be registered, c.f. |
| // common_types.h for the definition of |
| // CodecInst. |
| // |
| // Return value: |
| // -1 if failed to register the codec |
| // 0 if the codec registered successfully. |
| // |
| virtual int RegisterReceiveCodec(const CodecInst& receive_codec) = 0; |
| |
| // Register a decoder; call repeatedly to register multiple decoders. |df| is |
| // a decoder factory that returns an iSAC decoder; it will be called once if |
| // the decoder being registered is iSAC. |
| virtual int RegisterReceiveCodec( |
| const CodecInst& receive_codec, |
| rtc::FunctionView<std::unique_ptr<AudioDecoder>()> isac_factory) = 0; |
| |
| // Registers an external decoder. The name is only used to provide information |
| // back to the caller about the decoder. Hence, the name is arbitrary, and may |
| // be empty. |
| virtual int RegisterExternalReceiveCodec(int rtp_payload_type, |
| AudioDecoder* external_decoder, |
| int sample_rate_hz, |
| int num_channels, |
| const std::string& name) = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int32_t UnregisterReceiveCodec() |
| // Unregister the codec currently registered with a specific payload type |
| // from the list of possible receive codecs. |
| // |
| // Input: |
| // -payload_type : The number representing the payload type to |
| // unregister. |
| // |
| // Output: |
| // -1 if fails to unregister. |
| // 0 if the given codec is successfully unregistered. |
| // |
| virtual int UnregisterReceiveCodec( |
| uint8_t payload_type) = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int32_t ReceiveCodec() |
| // Get the codec associated with last received payload. |
| // |
| // Output: |
| // -curr_receive_codec : parameters of the codec associated with the last |
| // received payload, c.f. common_types.h for |
| // the definition of CodecInst. |
| // |
| // Return value: |
| // -1 if failed to retrieve the codec, |
| // 0 if the codec is successfully retrieved. |
| // |
| virtual int32_t ReceiveCodec(CodecInst* curr_receive_codec) const = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // rtc::Optional<SdpAudioFormat> ReceiveFormat() |
| // Get the format associated with last received payload. |
| // |
| // Return value: |
| // An SdpAudioFormat describing the format associated with the last |
| // received payload. |
| // An empty Optional if no payload has yet been received. |
| // |
| virtual rtc::Optional<SdpAudioFormat> ReceiveFormat() const = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int32_t IncomingPacket() |
| // Call this function to insert a parsed RTP packet into ACM. |
| // |
| // Inputs: |
| // -incoming_payload : received payload. |
| // -payload_len_bytes : the length of payload in bytes. |
| // -rtp_info : the relevant information retrieved from RTP |
| // header. |
| // |
| // Return value: |
| // -1 if failed to push in the payload |
| // 0 if payload is successfully pushed in. |
| // |
| virtual int32_t IncomingPacket(const uint8_t* incoming_payload, |
| const size_t payload_len_bytes, |
| const WebRtcRTPHeader& rtp_info) = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int32_t IncomingPayload() |
| // Call this API to push incoming payloads when there is no rtp-info. |
| // The rtp-info will be created in ACM. One usage for this API is when |
| // pre-encoded files are pushed in ACM |
| // |
| // Inputs: |
| // -incoming_payload : received payload. |
| // -payload_len_byte : the length, in bytes, of the received payload. |
| // -payload_type : the payload-type. This specifies which codec has |
| // to be used to decode the payload. |
| // -timestamp : send timestamp of the payload. ACM starts with |
| // a random value and increment it by the |
| // packet-size, which is given when the codec in |
| // question is registered by RegisterReceiveCodec(). |
| // Therefore, it is essential to have the timestamp |
| // if the frame-size differ from the registered |
| // value or if the incoming payload contains DTX |
| // packets. |
| // |
| // Return value: |
| // -1 if failed to push in the payload |
| // 0 if payload is successfully pushed in. |
| // |
| virtual int32_t IncomingPayload(const uint8_t* incoming_payload, |
| const size_t payload_len_byte, |
| const uint8_t payload_type, |
| const uint32_t timestamp = 0) = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int SetMinimumPlayoutDelay() |
| // Set a minimum for the playout delay, used for lip-sync. NetEq maintains |
| // such a delay unless channel condition yields to a higher delay. |
| // |
| // Input: |
| // -time_ms : minimum delay in milliseconds. |
| // |
| // Return value: |
| // -1 if failed to set the delay, |
| // 0 if the minimum delay is set. |
| // |
| virtual int SetMinimumPlayoutDelay(int time_ms) = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int SetMaximumPlayoutDelay() |
| // Set a maximum for the playout delay |
| // |
| // Input: |
| // -time_ms : maximum delay in milliseconds. |
| // |
| // Return value: |
| // -1 if failed to set the delay, |
| // 0 if the maximum delay is set. |
| // |
| virtual int SetMaximumPlayoutDelay(int time_ms) = 0; |
| |
| // TODO(kwiberg): Consider if this is needed anymore, now that voe::Channel |
| // doesn't use it. |
| // The shortest latency, in milliseconds, required by jitter buffer. This |
| // is computed based on inter-arrival times and playout mode of NetEq. The |
| // actual delay is the maximum of least-required-delay and the minimum-delay |
| // specified by SetMinumumPlayoutDelay() API. |
| // |
| virtual int LeastRequiredDelayMs() const = 0; |
| |
| // int32_t PlayoutTimestamp() |
| // The send timestamp of an RTP packet is associated with the decoded |
| // audio of the packet in question. This function returns the timestamp of |
| // the latest audio obtained by calling PlayoutData10ms(). |
| // |
| // Input: |
| // -timestamp : a reference to a uint32_t to receive the |
| // timestamp. |
| // Return value: |
| // 0 if the output is a correct timestamp. |
| // -1 if failed to output the correct timestamp. |
| // |
| RTC_DEPRECATED virtual int32_t PlayoutTimestamp(uint32_t* timestamp) = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int32_t PlayoutTimestamp() |
| // The send timestamp of an RTP packet is associated with the decoded |
| // audio of the packet in question. This function returns the timestamp of |
| // the latest audio obtained by calling PlayoutData10ms(), or empty if no |
| // valid timestamp is available. |
| // |
| virtual rtc::Optional<uint32_t> PlayoutTimestamp() = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int FilteredCurrentDelayMs() |
| // Returns the current total delay from NetEq (packet buffer and sync buffer) |
| // in ms, with smoothing applied to even out short-time fluctuations due to |
| // jitter. The packet buffer part of the delay is not updated during DTX/CNG |
| // periods. |
| // |
| virtual int FilteredCurrentDelayMs() const = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int32_t PlayoutData10Ms( |
| // Get 10 milliseconds of raw audio data for playout, at the given sampling |
| // frequency. ACM will perform a resampling if required. |
| // |
| // Input: |
| // -desired_freq_hz : the desired sampling frequency, in Hertz, of the |
| // output audio. If set to -1, the function returns |
| // the audio at the current sampling frequency. |
| // |
| // Output: |
| // -audio_frame : output audio frame which contains raw audio data |
| // and other relevant parameters, c.f. |
| // module_common_types.h for the definition of |
| // AudioFrame. |
| // -muted : if true, the sample data in audio_frame is not |
| // populated, and must be interpreted as all zero. |
| // |
| // Return value: |
| // -1 if the function fails, |
| // 0 if the function succeeds. |
| // |
| virtual int32_t PlayoutData10Ms(int32_t desired_freq_hz, |
| AudioFrame* audio_frame, |
| bool* muted) = 0; |
| |
| ///////////////////////////////////////////////////////////////////////////// |
| // Same as above, but without the muted parameter. This methods should not be |
| // used if enable_fast_accelerate was set to true in NetEq::Config. |
| // TODO(henrik.lundin) Remove this method when downstream dependencies are |
| // ready. |
| virtual int32_t PlayoutData10Ms(int32_t desired_freq_hz, |
| AudioFrame* audio_frame) = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // Codec specific |
| // |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int SetOpusApplication() |
| // Sets the intended application if current send codec is Opus. Opus uses this |
| // to optimize the encoding for applications like VOIP and music. Currently, |
| // two modes are supported: kVoip and kAudio. |
| // |
| // Input: |
| // - application : intended application. |
| // |
| // Return value: |
| // -1 if current send codec is not Opus or error occurred in setting the |
| // Opus application mode. |
| // 0 if the Opus application mode is successfully set. |
| // |
| virtual int SetOpusApplication(OpusApplicationMode application) = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int SetOpusMaxPlaybackRate() |
| // If current send codec is Opus, informs it about maximum playback rate the |
| // receiver will render. Opus can use this information to optimize the bit |
| // rate and increase the computation efficiency. |
| // |
| // Input: |
| // -frequency_hz : maximum playback rate in Hz. |
| // |
| // Return value: |
| // -1 if current send codec is not Opus or |
| // error occurred in setting the maximum playback rate, |
| // 0 if maximum bandwidth is set successfully. |
| // |
| virtual int SetOpusMaxPlaybackRate(int frequency_hz) = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // EnableOpusDtx() |
| // Enable the DTX, if current send codec is Opus. |
| // |
| // Return value: |
| // -1 if current send codec is not Opus or error occurred in enabling the |
| // Opus DTX. |
| // 0 if Opus DTX is enabled successfully. |
| // |
| virtual int EnableOpusDtx() = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int DisableOpusDtx() |
| // If current send codec is Opus, disables its internal DTX. |
| // |
| // Return value: |
| // -1 if current send codec is not Opus or error occurred in disabling DTX. |
| // 0 if Opus DTX is disabled successfully. |
| // |
| virtual int DisableOpusDtx() = 0; |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // statistics |
| // |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // int32_t GetNetworkStatistics() |
| // Get network statistics. Note that the internal statistics of NetEq are |
| // reset by this call. |
| // |
| // Input: |
| // -network_statistics : a structure that contains network statistics. |
| // |
| // Return value: |
| // -1 if failed to set the network statistics, |
| // 0 if statistics are set successfully. |
| // |
| virtual int32_t GetNetworkStatistics( |
| NetworkStatistics* network_statistics) = 0; |
| |
| // |
| // Enable NACK and set the maximum size of the NACK list. If NACK is already |
| // enable then the maximum NACK list size is modified accordingly. |
| // |
| // If the sequence number of last received packet is N, the sequence numbers |
| // of NACK list are in the range of [N - |max_nack_list_size|, N). |
| // |
| // |max_nack_list_size| should be positive (none zero) and less than or |
| // equal to |Nack::kNackListSizeLimit|. Otherwise, No change is applied and -1 |
| // is returned. 0 is returned at success. |
| // |
| virtual int EnableNack(size_t max_nack_list_size) = 0; |
| |
| // Disable NACK. |
| virtual void DisableNack() = 0; |
| |
| // |
| // Get a list of packets to be retransmitted. |round_trip_time_ms| is an |
| // estimate of the round-trip-time (in milliseconds). Missing packets which |
| // will be playout in a shorter time than the round-trip-time (with respect |
| // to the time this API is called) will not be included in the list. |
| // |
| // Negative |round_trip_time_ms| results is an error message and empty list |
| // is returned. |
| // |
| virtual std::vector<uint16_t> GetNackList( |
| int64_t round_trip_time_ms) const = 0; |
| |
| virtual void GetDecodingCallStatistics( |
| AudioDecodingCallStats* call_stats) const = 0; |
| |
| virtual ANAStats GetANAStats() const = 0; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_ |
