audio_coding: rename interface -> include

webrtc/modules/audio_coding/neteq/interface/neteq.h

Index: webrtc/modules/audio_coding/neteq/interface/neteq.h
diff --git a/webrtc/modules/audio_coding/neteq/interface/neteq.h b/webrtc/modules/audio_coding/neteq/interface/neteq.h
deleted file mode 100644
index 48e8fd5cdee61506dbb0af4fb48c4d6ecdd23c20..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/neteq/interface/neteq.h
+++ /dev/null
@@ -1,288 +0,0 @@
-/*
- *  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_NETEQ_INTERFACE_NETEQ_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_INTERFACE_NETEQ_H_
-
-#include <string.h>  // Provide access to size_t.
-
-#include <string>
-
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-// Forward declarations.
-struct WebRtcRTPHeader;
-
-struct NetEqNetworkStatistics {
-  uint16_t current_buffer_size_ms;  // Current jitter buffer size in ms.
-  uint16_t preferred_buffer_size_ms;  // Target buffer size in ms.
-  uint16_t jitter_peaks_found;  // 1 if adding extra delay due to peaky
-                                // jitter; 0 otherwise.
-  uint16_t packet_loss_rate;  // Loss rate (network + late) in Q14.
-  uint16_t packet_discard_rate;  // Late loss rate in Q14.
-  uint16_t expand_rate;  // Fraction (of original stream) of synthesized
-                         // audio inserted through expansion (in Q14).
-  uint16_t speech_expand_rate;  // Fraction (of original stream) of synthesized
-                                // speech inserted through expansion (in Q14).
-  uint16_t preemptive_rate;  // Fraction of data inserted through pre-emptive
-                             // expansion (in Q14).
-  uint16_t accelerate_rate;  // Fraction of data removed through acceleration
-                             // (in Q14).
-  uint16_t secondary_decoded_rate;  // Fraction of data coming from secondary
-                                    // decoding (in Q14).
-  int32_t clockdrift_ppm;  // Average clock-drift in parts-per-million
-                           // (positive or negative).
-  size_t added_zero_samples;  // Number of zero samples added in "off" mode.
-  // Statistics for packet waiting times, i.e., the time between a packet
-  // arrives until it is decoded.
-  int mean_waiting_time_ms;
-  int median_waiting_time_ms;
-  int min_waiting_time_ms;
-  int max_waiting_time_ms;
-};
-
-enum NetEqOutputType {
-  kOutputNormal,
-  kOutputPLC,
-  kOutputCNG,
-  kOutputPLCtoCNG,
-  kOutputVADPassive
-};
-
-enum NetEqPlayoutMode {
-  kPlayoutOn,
-  kPlayoutOff,
-  kPlayoutFax,
-  kPlayoutStreaming
-};
-
-// This is the interface class for NetEq.
-class NetEq {
- public:
-  enum BackgroundNoiseMode {
-    kBgnOn,    // Default behavior with eternal noise.
-    kBgnFade,  // Noise fades to zero after some time.
-    kBgnOff    // Background noise is always zero.
-  };
-
-  struct Config {
-    Config()
-        : sample_rate_hz(16000),
-          enable_audio_classifier(false),
-          max_packets_in_buffer(50),
-          // |max_delay_ms| has the same effect as calling SetMaximumDelay().
-          max_delay_ms(2000),
-          background_noise_mode(kBgnOff),
-          playout_mode(kPlayoutOn),
-          enable_fast_accelerate(false) {}
-
-    std::string ToString() const;
-
-    int sample_rate_hz;  // Initial value. Will change with input data.
-    bool enable_audio_classifier;
-    size_t max_packets_in_buffer;
-    int max_delay_ms;
-    BackgroundNoiseMode background_noise_mode;
-    NetEqPlayoutMode playout_mode;
-    bool enable_fast_accelerate;
-  };
-
-  enum ReturnCodes {
-    kOK = 0,
-    kFail = -1,
-    kNotImplemented = -2
-  };
-
-  enum ErrorCodes {
-    kNoError = 0,
-    kOtherError,
-    kInvalidRtpPayloadType,
-    kUnknownRtpPayloadType,
-    kCodecNotSupported,
-    kDecoderExists,
-    kDecoderNotFound,
-    kInvalidSampleRate,
-    kInvalidPointer,
-    kAccelerateError,
-    kPreemptiveExpandError,
-    kComfortNoiseErrorCode,
-    kDecoderErrorCode,
-    kOtherDecoderError,
-    kInvalidOperation,
-    kDtmfParameterError,
-    kDtmfParsingError,
-    kDtmfInsertError,
-    kStereoNotSupported,
-    kSampleUnderrun,
-    kDecodedTooMuch,
-    kFrameSplitError,
-    kRedundancySplitError,
-    kPacketBufferCorruption,
-    kSyncPacketNotAccepted
-  };
-
-  // Creates a new NetEq object, with parameters set in |config|. The |config|
-  // object will only have to be valid for the duration of the call to this
-  // method.
-  static NetEq* Create(const NetEq::Config& config);
-
-  virtual ~NetEq() {}
-
-  // Inserts a new packet into NetEq. The |receive_timestamp| is an indication
-  // of the time when the packet was received, and should be measured with
-  // the same tick rate as the RTP timestamp of the current payload.
-  // Returns 0 on success, -1 on failure.
-  virtual int InsertPacket(const WebRtcRTPHeader& rtp_header,
-                           const uint8_t* payload,
-                           size_t length_bytes,
-                           uint32_t receive_timestamp) = 0;
-
-  // Inserts a sync-packet into packet queue. Sync-packets are decoded to
-  // silence and are intended to keep AV-sync intact in an event of long packet
-  // losses when Video NACK is enabled but Audio NACK is not. Clients of NetEq
-  // might insert sync-packet when they observe that buffer level of NetEq is
-  // decreasing below a certain threshold, defined by the application.
-  // Sync-packets should have the same payload type as the last audio payload
-  // type, i.e. they cannot have DTMF or CNG payload type, nor a codec change
-  // can be implied by inserting a sync-packet.
-  // Returns kOk on success, kFail on failure.
-  virtual int InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
-                               uint32_t receive_timestamp) = 0;
-
-  // Instructs NetEq to deliver 10 ms of audio data. The data is written to
-  // |output_audio|, which can hold (at least) |max_length| elements.
-  // The number of channels that were written to the output is provided in
-  // the output variable |num_channels|, and each channel contains
-  // |samples_per_channel| elements. If more than one channel is written,
-  // the samples are interleaved.
-  // The speech type is written to |type|, if |type| is not NULL.
-  // Returns kOK on success, or kFail in case of an error.
-  virtual int GetAudio(size_t max_length, int16_t* output_audio,
-                       size_t* samples_per_channel, int* num_channels,
-                       NetEqOutputType* type) = 0;
-
-  // Associates |rtp_payload_type| with |codec| and stores the information in
-  // the codec database. Returns 0 on success, -1 on failure.
-  virtual int RegisterPayloadType(enum NetEqDecoder codec,
-                                  uint8_t rtp_payload_type) = 0;
-
-  // Provides an externally created decoder object |decoder| to insert in the
-  // decoder database. The decoder implements a decoder of type |codec| and
-  // associates it with |rtp_payload_type|. The decoder will produce samples
-  // at the rate |sample_rate_hz|. Returns kOK on success, kFail on failure.
-  virtual int RegisterExternalDecoder(AudioDecoder* decoder,
-                                      enum NetEqDecoder codec,
-                                      uint8_t rtp_payload_type,
-                                      int sample_rate_hz) = 0;
-
-  // Removes |rtp_payload_type| from the codec database. Returns 0 on success,
-  // -1 on failure.
-  virtual int RemovePayloadType(uint8_t rtp_payload_type) = 0;
-
-  // Sets a minimum delay in millisecond for packet buffer. The minimum is
-  // maintained unless a higher latency is dictated by channel condition.
-  // Returns true if the minimum is successfully applied, otherwise false is
-  // returned.
-  virtual bool SetMinimumDelay(int delay_ms) = 0;
-
-  // Sets a maximum delay in milliseconds for packet buffer. The latency will
-  // not exceed the given value, even required delay (given the channel
-  // conditions) is higher. Calling this method has the same effect as setting
-  // the |max_delay_ms| value in the NetEq::Config struct.
-  virtual bool SetMaximumDelay(int delay_ms) = 0;
-
-  // The smallest latency required. This is computed bases on inter-arrival
-  // time and internal NetEq logic. Note that in computing this latency none of
-  // the user defined limits (applied by calling setMinimumDelay() and/or
-  // SetMaximumDelay()) are applied.
-  virtual int LeastRequiredDelayMs() const = 0;
-
-  // Not implemented.
-  virtual int SetTargetDelay() = 0;
-
-  // Not implemented.
-  virtual int TargetDelay() = 0;
-
-  // Returns the current total delay (packet buffer and sync buffer) in ms.
-  virtual int CurrentDelayMs() const = 0;
-
-  // Sets the playout mode to |mode|.
-  // Deprecated. Set the mode in the Config struct passed to the constructor.
-  // TODO(henrik.lundin) Delete.
-  virtual void SetPlayoutMode(NetEqPlayoutMode mode) = 0;
-
-  // Returns the current playout mode.
-  // Deprecated.
-  // TODO(henrik.lundin) Delete.
-  virtual NetEqPlayoutMode PlayoutMode() const = 0;
-
-  // Writes the current network statistics to |stats|. The statistics are reset
-  // after the call.
-  virtual int NetworkStatistics(NetEqNetworkStatistics* stats) = 0;
-
-  // Writes the current RTCP statistics to |stats|. The statistics are reset
-  // and a new report period is started with the call.
-  virtual void GetRtcpStatistics(RtcpStatistics* stats) = 0;
-
-  // Same as RtcpStatistics(), but does not reset anything.
-  virtual void GetRtcpStatisticsNoReset(RtcpStatistics* stats) = 0;
-
-  // Enables post-decode VAD. When enabled, GetAudio() will return
-  // kOutputVADPassive when the signal contains no speech.
-  virtual void EnableVad() = 0;
-
-  // Disables post-decode VAD.
-  virtual void DisableVad() = 0;
-
-  // Gets the RTP timestamp for the last sample delivered by GetAudio().
-  // Returns true if the RTP timestamp is valid, otherwise false.
-  virtual bool GetPlayoutTimestamp(uint32_t* timestamp) = 0;
-
-  // Not implemented.
-  virtual int SetTargetNumberOfChannels() = 0;
-
-  // Not implemented.
-  virtual int SetTargetSampleRate() = 0;
-
-  // Returns the error code for the last occurred error. If no error has
-  // occurred, 0 is returned.
-  virtual int LastError() const = 0;
-
-  // Returns the error code last returned by a decoder (audio or comfort noise).
-  // When LastError() returns kDecoderErrorCode or kComfortNoiseErrorCode, check
-  // this method to get the decoder's error code.
-  virtual int LastDecoderError() = 0;
-
-  // Flushes both the packet buffer and the sync buffer.
-  virtual void FlushBuffers() = 0;
-
-  // Current usage of packet-buffer and it's limits.
-  virtual void PacketBufferStatistics(int* current_num_packets,
-                                      int* max_num_packets) const = 0;
-
-  // Get sequence number and timestamp of the latest RTP.
-  // This method is to facilitate NACK.
-  virtual int DecodedRtpInfo(int* sequence_number,
-                             uint32_t* timestamp) const = 0;
-
- protected:
-  NetEq() {}
-
- private:
-  RTC_DISALLOW_COPY_AND_ASSIGN(NetEq);
-};
-
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INTERFACE_NETEQ_H_