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diff --git a/webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h b/webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h
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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_INCLUDE_OPUS_INTERFACE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INCLUDE_OPUS_INTERFACE_H_
-
-#include <stddef.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;
-
-/****************************************************************************
- * WebRtcOpus_EncoderCreate(...)
- *
- * This function create an Opus encoder.
- *
- * Input:
- * - channels : number of channels.
- * - application : 0 - VOIP applications.
- * Favor speech intelligibility.
- * 1 - Audio applications.
- * Favor faithfulness to the original input.
- *
- * Output:
- * - inst : a pointer to Encoder context that is created
- * if success.
- *
- * Return value : 0 - Success
- * -1 - Error
- */
-int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst,
- int32_t channels,
- int32_t application);
-
-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
- */
-int WebRtcOpus_Encode(OpusEncInst* inst,
- const int16_t* audio_in,
- size_t samples,
- size_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_SetMaxPlaybackRate(...)
- *
- * Configures the maximum playback rate for encoding. Due to hardware
- * limitations, the receiver may render audio up to a playback rate. Opus
- * encoder can use this information to optimize for network usage and encoding
- * complexity. This will affect the audio bandwidth in the coded audio. However,
- * the input/output sample rate is not affected.
- *
- * Input:
- * - inst : Encoder context
- * - frequency_hz : Maximum playback rate in Hz.
- * This parameter can take any value. The relation
- * between the value and the Opus internal mode is
- * as following:
- * frequency_hz <= 8000 narrow band
- * 8000 < frequency_hz <= 12000 medium band
- * 12000 < frequency_hz <= 16000 wide band
- * 16000 < frequency_hz <= 24000 super wide band
- * frequency_hz > 24000 full band
- * Return value : 0 - Success
- * -1 - Error
- */
-int16_t WebRtcOpus_SetMaxPlaybackRate(OpusEncInst* inst, int32_t frequency_hz);
-
-/* 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_EnableDtx()
- *
- * This function enables Opus internal DTX for encoding.
- *
- * Input:
- * - inst : Encoder context
- *
- * Return value : 0 - Success
- * -1 - Error
- */
-int16_t WebRtcOpus_EnableDtx(OpusEncInst* inst);
-
-/****************************************************************************
- * WebRtcOpus_DisableDtx()
- *
- * This function disables Opus internal DTX for encoding.
- *
- * Input:
- * - inst : Encoder context
- *
- * Return value : 0 - Success
- * -1 - Error
- */
-int16_t WebRtcOpus_DisableDtx(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
- */
-void WebRtcOpus_DecoderInit(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
- */
-int WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
- size_t encoded_bytes, int16_t* decoded,
- int16_t* audio_type);
-
-/****************************************************************************
- * WebRtcOpus_DecodePlc(...)
- *
- * 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
- */
-int WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
- int 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
- */
-int WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
- size_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 per
- * channel.
- */
-int WebRtcOpus_DurationEst(OpusDecInst* inst,
- const uint8_t* payload,
- size_t payload_length_bytes);
-
-/****************************************************************************
- * WebRtcOpus_PlcDuration(...)
- *
- * This function calculates the duration of a frame returned by packet loss
- * concealment (PLC).
- *
- * Input:
- * - inst : Decoder context
- *
- * Return value : The duration of a frame returned by PLC, in
- * samples per channel.
- */
-int WebRtcOpus_PlcDuration(OpusDecInst* inst);
-
-/* 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 per channel.
- * 0 - No FEC data in the packet.
- */
-int WebRtcOpus_FecDurationEst(const uint8_t* payload,
- size_t 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,
- size_t payload_length_bytes);
-
-#ifdef __cplusplus
-} // extern "C"
-#endif
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INCLUDE_OPUS_INCLUDE_H_