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Support for Signal calls.

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// FREEBIE
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Moxie Marlinspike
2015-09-09 13:54:29 -07:00
parent 3d4ae60d81
commit d83a3d71bc
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jni/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h Normal file
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/*
* 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_CODECS_OPUS_INTERFACE_OPUS_INTERFACE_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_OPUS_INTERFACE_H_
#include "webrtc/typedefs.h"
#ifdef __cplusplus
extern "C" {
#endif
// Opaque wrapper types for the codec state.
typedef struct WebRtcOpusEncInst OpusEncInst;
typedef struct WebRtcOpusDecInst OpusDecInst;
int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst, int32_t channels);
int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst);
/****************************************************************************
* WebRtcOpus_Encode(...)
*
* This function encodes audio as a series of Opus frames and inserts
* it into a packet. Input buffer can be any length.
*
* Input:
* - inst : Encoder context
* - audio_in : Input speech data buffer
* - samples : Samples per channel in audio_in
* - length_encoded_buffer : Output buffer size
*
* Output:
* - encoded : Output compressed data buffer
*
* Return value : >0 - Length (in bytes) of coded data
* -1 - Error
*/
int16_t WebRtcOpus_Encode(OpusEncInst* inst, int16_t* audio_in, int16_t samples,
int16_t length_encoded_buffer, uint8_t* encoded);
/****************************************************************************
* WebRtcOpus_SetBitRate(...)
*
* This function adjusts the target bitrate of the encoder.
*
* Input:
* - inst : Encoder context
* - rate : New target bitrate
*
* Return value : 0 - Success
* -1 - Error
*/
int16_t WebRtcOpus_SetBitRate(OpusEncInst* inst, int32_t rate);
/****************************************************************************
* WebRtcOpus_SetPacketLossRate(...)
*
* This function configures the encoder's expected packet loss percentage.
*
* Input:
* - inst : Encoder context
* - loss_rate : loss percentage in the range 0-100, inclusive.
* Return value : 0 - Success
* -1 - Error
*/
int16_t WebRtcOpus_SetPacketLossRate(OpusEncInst* inst, int32_t loss_rate);
/****************************************************************************
* WebRtcOpus_SetMaxBandwidth(...)
*
* Configures the maximum bandwidth for encoding. This can be taken as a hint
* about the maximum output bandwidth that the receiver is capable to render,
* due to hardware limitations. Sending signals with higher audio bandwidth
* results in higher than necessary network usage and encoding complexity.
*
* Input:
* - inst : Encoder context
* - bandwidth : Maximum encoding bandwidth in Hz.
* This parameter can take any value, but values
* other than Opus typical bandwidths: 4000, 6000,
* 8000, 12000, and 20000 will be rounded up (values
* greater than 20000 will be rounded down) to
* these values.
* Return value : 0 - Success
* -1 - Error
*/
int16_t WebRtcOpus_SetMaxBandwidth(OpusEncInst* inst, int32_t bandwidth);
/* TODO(minyue): Check whether an API to check the FEC and the packet loss rate
* is needed. It might not be very useful since there are not many use cases and
* the caller can always maintain the states. */
/****************************************************************************
* WebRtcOpus_EnableFec()
*
* This function enables FEC for encoding.
*
* Input:
* - inst : Encoder context
*
* Return value : 0 - Success
* -1 - Error
*/
int16_t WebRtcOpus_EnableFec(OpusEncInst* inst);
/****************************************************************************
* WebRtcOpus_DisableFec()
*
* This function disables FEC for encoding.
*
* Input:
* - inst : Encoder context
*
* Return value : 0 - Success
* -1 - Error
*/
int16_t WebRtcOpus_DisableFec(OpusEncInst* inst);
/*
* WebRtcOpus_SetComplexity(...)
*
* This function adjusts the computational complexity. The effect is the same as
* calling the complexity setting of Opus as an Opus encoder related CTL.
*
* Input:
* - inst : Encoder context
* - complexity : New target complexity (0-10, inclusive)
*
* Return value : 0 - Success
* -1 - Error
*/
int16_t WebRtcOpus_SetComplexity(OpusEncInst* inst, int32_t complexity);
int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels);
int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst);
/****************************************************************************
* WebRtcOpus_DecoderChannels(...)
*
* This function returns the number of channels created for Opus decoder.
*/
int WebRtcOpus_DecoderChannels(OpusDecInst* inst);
/****************************************************************************
* WebRtcOpus_DecoderInit(...)
*
* This function resets state of the decoder.
*
* Input:
* - inst : Decoder context
*
* Return value : 0 - Success
* -1 - Error
*/
int16_t WebRtcOpus_DecoderInitNew(OpusDecInst* inst);
int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst);
int16_t WebRtcOpus_DecoderInitSlave(OpusDecInst* inst);
/****************************************************************************
* WebRtcOpus_Decode(...)
*
* This function decodes an Opus packet into one or more audio frames at the
* ACM interface's sampling rate (32 kHz).
*
* Input:
* - inst : Decoder context
* - encoded : Encoded data
* - encoded_bytes : Bytes in encoded vector
*
* Output:
* - decoded : The decoded vector
* - audio_type : 1 normal, 2 CNG (for Opus it should
* always return 1 since we're not using Opus's
* built-in DTX/CNG scheme)
*
* Return value : >0 - Samples per channel in decoded vector
* -1 - Error
*/
int16_t WebRtcOpus_DecodeNew(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type);
int16_t WebRtcOpus_Decode(OpusDecInst* inst, const int16_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type);
int16_t WebRtcOpus_DecodeSlave(OpusDecInst* inst, const int16_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type);
/****************************************************************************
* WebRtcOpus_DecodePlc(...)
* TODO(tlegrand): Remove master and slave functions when NetEq4 is in place.
* WebRtcOpus_DecodePlcMaster(...)
* WebRtcOpus_DecodePlcSlave(...)
*
* This function processes PLC for opus frame(s).
* Input:
* - inst : Decoder context
* - number_of_lost_frames : Number of PLC frames to produce
*
* Output:
* - decoded : The decoded vector
*
* Return value : >0 - number of samples in decoded PLC vector
* -1 - Error
*/
int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
int16_t number_of_lost_frames);
int16_t WebRtcOpus_DecodePlcMaster(OpusDecInst* inst, int16_t* decoded,
int16_t number_of_lost_frames);
int16_t WebRtcOpus_DecodePlcSlave(OpusDecInst* inst, int16_t* decoded,
int16_t number_of_lost_frames);
/****************************************************************************
* WebRtcOpus_DecodeFec(...)
*
* This function decodes the FEC data from an Opus packet into one or more audio
* frames at the ACM interface's sampling rate (32 kHz).
*
* Input:
* - inst : Decoder context
* - encoded : Encoded data
* - encoded_bytes : Bytes in encoded vector
*
* Output:
* - decoded : The decoded vector (previous frame)
*
* Return value : >0 - Samples per channel in decoded vector
* 0 - No FEC data in the packet
* -1 - Error
*/
int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type);
/****************************************************************************
* WebRtcOpus_DurationEst(...)
*
* This function calculates the duration of an opus packet.
* Input:
* - inst : Decoder context
* - payload : Encoded data pointer
* - payload_length_bytes : Bytes of encoded data
*
* Return value : The duration of the packet, in samples.
*/
int WebRtcOpus_DurationEst(OpusDecInst* inst,
const uint8_t* payload,
int payload_length_bytes);
/* TODO(minyue): Check whether it is needed to add a decoder context to the
* arguments, like WebRtcOpus_DurationEst(...). In fact, the packet itself tells
* the duration. The decoder context in WebRtcOpus_DurationEst(...) is not used.
* So it may be advisable to remove it from WebRtcOpus_DurationEst(...). */
/****************************************************************************
* WebRtcOpus_FecDurationEst(...)
*
* This function calculates the duration of the FEC data within an opus packet.
* Input:
* - payload : Encoded data pointer
* - payload_length_bytes : Bytes of encoded data
*
* Return value : >0 - The duration of the FEC data in the
* packet in samples.
* 0 - No FEC data in the packet.
*/
int WebRtcOpus_FecDurationEst(const uint8_t* payload,
int payload_length_bytes);
/****************************************************************************
* WebRtcOpus_PacketHasFec(...)
*
* This function detects if an opus packet has FEC.
* Input:
* - payload : Encoded data pointer
* - payload_length_bytes : Bytes of encoded data
*
* Return value : 0 - the packet does NOT contain FEC.
* 1 - the packet contains FEC.
*/
int WebRtcOpus_PacketHasFec(const uint8_t* payload,
int payload_length_bytes);
#ifdef __cplusplus
} // extern "C"
#endif
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_OPUS_INTERFACE_H_