| /* |
| * 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_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_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_ |
