webrtc/webrtc/modules/audio_coding/main/interface/audio_coding_module.h

/*
 *  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_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_

#include <vector>

#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
#include "webrtc/modules/interface/module.h"
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/typedefs.h"

namespace webrtc {

// forward declarations
struct CodecInst;
struct WebRtcRTPHeader;
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 inband Dtmf detection
class AudioCodingFeedback {
 public:
  virtual ~AudioCodingFeedback() {}

  virtual int32_t IncomingDtmf(const uint8_t digit_dtmf,
                               const bool end) = 0;
};

// Callback class used for reporting VAD decision
class ACMVADCallback {
 public:
  virtual ~ACMVADCallback() {}

  virtual int32_t InFrameType(int16_t frameType) = 0;
};

// Callback class used for reporting receiver statistics
class ACMVQMonCallback {
 public:
  virtual ~ACMVQMonCallback() {}

  virtual int32_t NetEqStatistics(
      const int32_t id,  // current ACM id
      const uint16_t MIUsValid,  // valid voice duration in ms
      const uint16_t MIUsReplaced,  // concealed voice duration in ms
      const uint8_t eventFlags,  // concealed voice flags
      const uint16_t delayMS) = 0;  // average delay in ms
};

class AudioCodingModule {
 protected:
  AudioCodingModule() {}

 public:
  struct Config {
    Config()
        : id(0),
          neteq_config(),
          clock(Clock::GetRealTimeClock()) {}

    int id;
    NetEq::Config neteq_config;
    Clock* clock;
  };

  ///////////////////////////////////////////////////////////////////////////
  // 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);
  virtual ~AudioCodingModule() {};

  ///////////////////////////////////////////////////////////////////////////
  //   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, int 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,
                             int 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 InitializeSender()
  // Any encoder-related state of ACM will be initialized to the
  // same state when ACM is created. This will not interrupt or
  // effect decoding functionality of ACM. ACM will lose all the
  // encoding-related settings by calling this function.
  // For instance, a send codec has to be registered again.
  //
  // Return value:
  //   -1 if failed to initialize,
  //    0 if succeeded.
  //
  virtual int32_t InitializeSender() = 0;

  ///////////////////////////////////////////////////////////////////////////
  // int32_t ResetEncoder()
  // This API resets the states of encoder. All the encoder settings, such as
  // send-codec or VAD/DTX, will be preserved.
  //
  // Return value:
  //   -1 if failed to initialize,
  //    0 if succeeded.
  //
  virtual int32_t ResetEncoder() = 0;

  ///////////////////////////////////////////////////////////////////////////
  // 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;

  ///////////////////////////////////////////////////////////////////////////
  // int32_t SendCodec()
  // Get parameters for the codec currently registered as send codec.
  //
  // Output:
  //   -current_send_codec          : parameters of the send codec.
  //
  // Return value:
  //   -1 if failed to get send codec,
  //    0 if succeeded.
  //
  virtual int32_t SendCodec(CodecInst* current_send_codec) 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;

  ///////////////////////////////////////////////////////////////////////////
  // int32_t Bitrate()
  // Get encoding bit-rate in bits per second.
  //
  // Return value:
  //   positive; encoding rate in bits/sec,
  //   -1 if an error is happened.
  //
  virtual int32_t SendBitrate() const = 0;

  ///////////////////////////////////////////////////////////////////////////
  // int32_t SetReceivedEstimatedBandwidth()
  // Set available bandwidth [bits/sec] of the up-link channel.
  // This information is used for traffic shaping, and is currently only
  // supported if iSAC is the send codec.
  //
  // Input:
  //   -bw                 : bandwidth in bits/sec estimated for
  //                         up-link.
  // Return value
  //   -1 if error occurred in setting the bandwidth,
  //    0 bandwidth is set successfully.
  //
  // TODO(henrik.lundin) Unused. Remove?
  virtual int32_t SetReceivedEstimatedBandwidth(
      const int32_t bw) = 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.
  //
  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 ReplaceInternalDTXWithWebRtc()
  // Used to replace codec internal DTX scheme with WebRtc.
  //
  // Input:
  //   -use_webrtc_dtx     : if false (default) the codec built-in DTX/VAD
  //                         scheme is used, otherwise the internal DTX is
  //                         replaced with WebRtc DTX/VAD.
  //
  // Return value:
  //   -1 if failed to replace codec internal DTX with WebRtc,
  //    0 if succeeded.
  //
  virtual int32_t ReplaceInternalDTXWithWebRtc(
      const bool use_webrtc_dtx = false) = 0;

  ///////////////////////////////////////////////////////////////////////////
  // int32_t IsInternalDTXReplacedWithWebRtc()
  // Get status if the codec internal DTX is replaced with WebRtc DTX.
  // This should always be true if codec does not have an internal DTX.
  //
  // Output:
  //   -uses_webrtc_dtx    : is set to true if the codec internal DTX is
  //                         replaced with WebRtc DTX/VAD, otherwise it is set
  //                         to false.
  //
  // Return value:
  //   -1 if failed to determine if codec internal DTX is replaced with WebRtc,
  //    0 if succeeded.
  //
  virtual int32_t IsInternalDTXReplacedWithWebRtc(
      bool* uses_webrtc_dtx) = 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 ResetDecoder()
  // This API resets the states of decoders. ACM will not lose any
  // decoder-related settings, such as registered codecs.
  //
  // Return value:
  //   -1 if failed to initialize,
  //    0 if succeeded.
  //
  virtual int32_t ResetDecoder() = 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;

  ///////////////////////////////////////////////////////////////////////////
  // 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 int32_t RegisterReceiveCodec(
      const CodecInst& receive_codec) = 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;

  ///////////////////////////////////////////////////////////////////////////
  // 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;

  //
  // 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 SetDtmfPlayoutStatus()
  // Configure DTMF playout, i.e. whether out-of-band
  // DTMF tones are played or not.
  //
  // Input:
  //   -enable             : if true to enable playout out-of-band DTMF tones,
  //                         false to disable.
  //
  // Return value:
  //   -1 if the method fails, e.g. DTMF playout is not supported.
  //    0 if the status is set successfully.
  //
  virtual int32_t SetDtmfPlayoutStatus(const bool enable) = 0;

  ///////////////////////////////////////////////////////////////////////////
  // bool DtmfPlayoutStatus()
  // Get Dtmf playout status.
  //
  // Return value:
  //   true if out-of-band Dtmf tones are played,
  //   false if playout of Dtmf tones is disabled.
  //
  virtual bool DtmfPlayoutStatus() 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.
  //
  // TODO(tlegrand): Change function to return the timestamp.
  virtual int32_t PlayoutTimestamp(uint32_t* timestamp) = 0;

  ///////////////////////////////////////////////////////////////////////////
  // int32_t DecoderEstimatedBandwidth()
  // Get the estimate of the Bandwidth, in bits/second, based on the incoming
  // stream. This API is useful in one-way communication scenarios, where
  // the bandwidth information is sent in an out-of-band fashion.
  // Currently only supported if iSAC is registered as a receiver.
  //
  // Return value:
  //   >0 bandwidth in bits/second.
  //   -1 if failed to get a bandwidth estimate.
  //
  virtual int32_t DecoderEstimatedBandwidth() const = 0;

  ///////////////////////////////////////////////////////////////////////////
  // int32_t SetPlayoutMode()
  // Call this API to set the playout mode. Playout mode could be optimized
  // for i) voice, ii) FAX or iii) streaming. In Voice mode, NetEQ is
  // optimized to deliver highest audio quality while maintaining a minimum
  // delay. In FAX mode, NetEQ is optimized to have few delay changes as
  // possible and maintain a constant delay, perhaps large relative to voice
  // mode, to avoid PLC. In streaming mode, we tolerate a little more delay
  // to achieve better jitter robustness.
  //
  // Input:
  //   -mode               : playout mode. Possible inputs are:
  //                         "voice",
  //                         "fax" and
  //                         "streaming".
  //
  // Return value:
  //   -1 if failed to set the mode,
  //    0 if succeeding.
  //
  virtual int32_t SetPlayoutMode(const AudioPlayoutMode mode) = 0;

  ///////////////////////////////////////////////////////////////////////////
  // AudioPlayoutMode PlayoutMode()
  // Get playout mode, i.e. whether it is speech, FAX or streaming. See
  // audio_coding_module_typedefs.h for definition of AudioPlayoutMode.
  //
  // Return value:
  //   voice:       is for voice output,
  //   fax:         a mode that is optimized for receiving FAX signals.
  //                In this mode NetEq tries to maintain a constant high
  //                delay to avoid PLC if possible.
  //   streaming:   a mode that is suitable for streaming. In this mode we
  //                accept longer delay to improve jitter robustness.
  //
  virtual AudioPlayoutMode PlayoutMode() 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.
  //
  // Return value:
  //   -1 if the function fails,
  //    0 if the function succeeds.
  //
  virtual int32_t PlayoutData10Ms(int32_t desired_freq_hz,
                                        AudioFrame* audio_frame) = 0;

  ///////////////////////////////////////////////////////////////////////////
  //   Codec specific
  //

  ///////////////////////////////////////////////////////////////////////////
  // int32_t SetISACMaxRate()
  // Set the maximum instantaneous rate of iSAC. For a payload of B bits
  // with a frame-size of T sec the instantaneous rate is B/T bits per
  // second. Therefore, (B/T < |max_rate_bps|) and
  // (B < |max_payload_len_bytes| * 8) are always satisfied for iSAC payloads,
  // c.f SetISACMaxPayloadSize().
  //
  // Input:
  //   -max_rate_bps       : maximum instantaneous bit-rate given in bits/sec.
  //
  // Return value:
  //   -1 if failed to set the maximum rate.
  //    0 if the maximum rate is set successfully.
  //
  virtual int SetISACMaxRate(int max_rate_bps) = 0;

  ///////////////////////////////////////////////////////////////////////////
  // int32_t SetISACMaxPayloadSize()
  // Set the maximum payload size of iSAC packets. No iSAC payload,
  // regardless of its frame-size, may exceed the given limit. For
  // an iSAC payload of size B bits and frame-size T seconds we have;
  // (B < |max_payload_len_bytes| * 8) and (B/T < |max_rate_bps|), c.f.
  // SetISACMaxRate().
  //
  // Input:
  //   -max_payload_len_bytes : maximum payload size in bytes.
  //
  // Return value:
  //   -1 if failed to set the maximum  payload-size.
  //    0 if the given length is set successfully.
  //
  virtual int SetISACMaxPayloadSize(int max_payload_len_bytes) = 0;

  ///////////////////////////////////////////////////////////////////////////
  // int32_t ConfigISACBandwidthEstimator()
  // Call this function to configure the bandwidth estimator of ISAC.
  // During the adaptation of bit-rate, iSAC automatically adjusts the
  // frame-size (either 30 or 60 ms) to save on RTP header. The initial
  // frame-size can be specified by the first argument. The configuration also
  // regards the initial estimate of bandwidths. The estimator starts from
  // this point and converges to the actual bottleneck. This is given by the
  // second parameter. Furthermore, it is also possible to control the
  // adaptation of frame-size. This is specified by the last parameter.
  //
  // Input:
  //   -init_frame_size_ms : initial frame-size in milliseconds. For iSAC-wb
  //                         30 ms and 60 ms (default) are acceptable values,
  //                         and for iSAC-swb 30 ms is the only acceptable
  //                         value. Zero indicates default value.
  //   -init_rate_bps      : initial estimate of the bandwidth. Values
  //                         between 10000 and 58000 are acceptable.
  //   -enforce_srame_size : if true, the frame-size will not be adapted.
  //
  // Return value:
  //   -1 if failed to configure the bandwidth estimator,
  //    0 if the configuration was successfully applied.
  //
  virtual int32_t ConfigISACBandwidthEstimator(
      int init_frame_size_ms,
      int init_rate_bps,
      bool enforce_frame_size = false) = 0;

  ///////////////////////////////////////////////////////////////////////////
  // int SetOpusApplication()
  // Sets the intended application for the Opus encoder. Opus uses this to
  // optimize the encoding for applications like VOIP and music.
  //
  // Input:
  //   - application      : intended application.
  //
  // Return value:
  //   -1 if failed or on codecs other than Opus.
  //    0 if succeeded.
  //
  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 maximum bandwidth is set successfully.
  //
  virtual int SetOpusMaxPlaybackRate(int frequency_hz) = 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;

  //
  // Set an initial delay for playout.
  // An initial delay yields ACM playout silence until equivalent of |delay_ms|
  // audio payload is accumulated in NetEq jitter. Thereafter, ACM pulls audio
  // from NetEq in its regular fashion, and the given delay is maintained
  // through out the call, unless channel conditions yield to a higher jitter
  // buffer delay.
  //
  // Input:
  //   -delay_ms           : delay in milliseconds.
  //
  // Return values:
  //   -1 if failed to set the delay.
  //    0 if delay is set successfully.
  //
  virtual int SetInitialPlayoutDelay(int delay_ms) = 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;
};

class AudioEncoder;
class ReceiverInfo;

class AudioCoding {
 public:
  struct Config {
    Config()
        : neteq_config(),
          clock(Clock::GetRealTimeClock()),
          transport(NULL),
          vad_callback(NULL),
          play_dtmf(true),
          initial_playout_delay_ms(0),
          playout_channels(1),
          playout_frequency_hz(32000) {}

    AudioCodingModule::Config ToOldConfig() const {
      AudioCodingModule::Config old_config;
      old_config.id = 0;
      old_config.neteq_config = neteq_config;
      old_config.clock = clock;
      return old_config;
    }

    NetEq::Config neteq_config;
    Clock* clock;
    AudioPacketizationCallback* transport;
    ACMVADCallback* vad_callback;
    bool play_dtmf;
    int initial_playout_delay_ms;
    int playout_channels;
    int playout_frequency_hz;
  };

  static AudioCoding* Create(const Config& config);
  virtual ~AudioCoding() {};

  // Registers a codec, specified by |send_codec|, as sending codec.
  // This API can be called multiple times. The last codec registered overwrites
  // the previous ones. Returns true if successful, false if not.
  //
  // 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.
  virtual bool RegisterSendCodec(AudioEncoder* send_codec) = 0;

  // Temporary solution to be used during refactoring:
  // |encoder_type| should be from the anonymous enum in acm2::ACMCodecDB.
  virtual bool RegisterSendCodec(int encoder_type,
                                 uint8_t payload_type,
                                 int frame_size_samples = 0) = 0;

  // Returns the encoder object currently in use. This is the same as the
  // codec that was registered in the latest call to RegisterSendCodec().
  virtual const AudioEncoder* GetSenderInfo() const = 0;

  // Temporary solution to be used during refactoring.
  virtual const CodecInst* GetSenderCodecInst() = 0;

  // Adds 10 ms of raw (PCM) audio data to the encoder. If the sampling
  // frequency of the audio does not match the sampling frequency of the
  // current encoder, ACM will resample the audio.
  //
  // Return value:
  //      0   successfully added the frame.
  //     -1   some error occurred and data is not added.
  //   < -1   to add the frame to the buffer n samples had to be
  //          overwritten, -n is the return value in this case.
  // TODO(henrik.lundin): Make a better design for the return values. This one
  // is just a copy of the old API.
  virtual int Add10MsAudio(const AudioFrame& audio_frame) = 0;

  // Returns a combined info about the currently used decoder(s).
  virtual const ReceiverInfo* GetReceiverInfo() const = 0;

  // Registers a codec, specified by |receive_codec|, as receiving codec.
  // This API can be called multiple times. If registering with a payload type
  // that was already registered in a previous call, the latest call will
  // override previous calls. Returns true if successful, false if not.
  virtual bool RegisterReceiveCodec(AudioDecoder* receive_codec) = 0;

  // Temporary solution:
  // |decoder_type| should be from the anonymous enum in acm2::ACMCodecDB.
  virtual bool RegisterReceiveCodec(int decoder_type, uint8_t payload_type) = 0;

  // The following two methods both inserts a new packet to the receiver.
  // InsertPacket takes an RTP header input in |rtp_info|, while InsertPayload
  // only requires a payload type and a timestamp. The latter assumes that the
  // payloads come in the right order, and without any losses. In both cases,
  // |incoming_payload| contains the RTP payload after the RTP header. Return
  // true if successful, false if not.
  virtual bool InsertPacket(const uint8_t* incoming_payload,
                            size_t payload_len_bytes,
                            const WebRtcRTPHeader& rtp_info) = 0;

  // TODO(henrik.lundin): Remove this method?
  virtual bool InsertPayload(const uint8_t* incoming_payload,
                             size_t payload_len_byte,
                             uint8_t payload_type,
                             uint32_t timestamp) = 0;

  // These two methods set a minimum and maximum jitter buffer delay in
  // milliseconds. The pupose is mainly to adjust the delay to synchronize
  // audio and video. The preferred jitter buffer size, computed by NetEq based
  // on the current channel conditions, is clamped from below and above by these
  // two methods. The given delay limits must be non-negative, less than
  // 10000 ms, and the minimum must be strictly smaller than the maximum.
  // Further, the maximum must be at lest one frame duration. If these
  // conditions are not met, false is returned. Giving the value 0 effectively
  // unsets the minimum or maximum delay limits.
  // Note that calling these methods is optional. If not called, NetEq will
  // determine the optimal buffer size based on the network conditions.
  virtual bool SetMinimumPlayoutDelay(int time_ms) = 0;

  virtual bool SetMaximumPlayoutDelay(int time_ms) = 0;

  // Returns the current value of the jitter buffer's preferred latency. This
  // is computed based on inter-arrival times and playout mode of NetEq. The
  // actual target delay is this value clamped from below and above by the
  // values specified through SetMinimumPlayoutDelay() and
  // SetMaximumPlayoutDelay(), respectively, if provided.
  // TODO(henrik.lundin) Rename to PreferredDelayMs?
  virtual int LeastRequiredDelayMs() const = 0;

  // 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 delivered by Get10MsAudio(). Returns false if no timestamp
  // can be provided, true otherwise.
  virtual bool PlayoutTimestamp(uint32_t* timestamp) = 0;

  // Delivers 10 ms of audio in |audio_frame|. Returns true if successful,
  // false otherwise.
  virtual bool Get10MsAudio(AudioFrame* audio_frame) = 0;

  // Returns the network statistics. Note that the internal statistics of NetEq
  // are reset by this call. Returns true if successful, false otherwise.
  virtual bool GetNetworkStatistics(NetworkStatistics* network_statistics) = 0;

  // Enables NACK and sets the maximum size of the NACK list. If NACK is already
  // enabled then the maximum NACK list size is modified accordingly. Returns
  // true if successful, false otherwise.
  //
  // 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 and less than or equal to
  // |Nack::kNackListSizeLimit|.
  virtual bool EnableNack(size_t max_nack_list_size) = 0;

  // Disables NACK.
  virtual void DisableNack() = 0;


  // Temporary solution to be used during refactoring.
  // If DTX is enabled and 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 sent,
  // 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.
  //
  // Return true if successful, false otherwise.
  virtual bool SetVad(bool enable_dtx,
                      bool enable_vad,
                      ACMVADMode vad_mode) = 0;

  // Returns a list of packets to request retransmission of.
  // |round_trip_time_ms| is an estimate of the round-trip-time (in
  // milliseconds). Missing packets which will be decoded sooner than the
  // round-trip-time (with respect to the time this API is called) will not be
  // included in the list.
  // |round_trip_time_ms| must be non-negative.
  virtual std::vector<uint16_t> GetNackList(int round_trip_time_ms) const = 0;

  // Returns the timing statistics for calls to Get10MsAudio.
  virtual void GetDecodingCallStatistics(
      AudioDecodingCallStats* call_stats) const = 0;
};

}  // namespace webrtc

#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_