| /* |
| * 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. |
| */ |
| |
| #include "modules/audio_coding/codecs/opus/opus_interface.h" |
| |
| #include "rtc_base/checks.h" |
| |
| #include <stdlib.h> |
| #include <string.h> |
| |
| enum { |
| #if WEBRTC_OPUS_SUPPORT_120MS_PTIME |
| /* Maximum supported frame size in WebRTC is 120 ms. */ |
| kWebRtcOpusMaxEncodeFrameSizeMs = 120, |
| #else |
| /* Maximum supported frame size in WebRTC is 60 ms. */ |
| kWebRtcOpusMaxEncodeFrameSizeMs = 60, |
| #endif |
| |
| /* The format allows up to 120 ms frames. Since we don't control the other |
| * side, we must allow for packets of that size. NetEq is currently limited |
| * to 60 ms on the receive side. */ |
| kWebRtcOpusMaxDecodeFrameSizeMs = 120, |
| }; |
| |
| static int FrameSizePerChannel(int frame_size_ms, int sample_rate_hz) { |
| RTC_DCHECK_GT(frame_size_ms, 0); |
| RTC_DCHECK_EQ(frame_size_ms % 10, 0); |
| RTC_DCHECK_GT(sample_rate_hz, 0); |
| RTC_DCHECK_EQ(sample_rate_hz % 1000, 0); |
| return frame_size_ms * (sample_rate_hz / 1000); |
| } |
| |
| // Maximum sample count per channel. |
| static int MaxFrameSizePerChannel(int sample_rate_hz) { |
| return FrameSizePerChannel(kWebRtcOpusMaxDecodeFrameSizeMs, sample_rate_hz); |
| } |
| |
| // Default sample count per channel. |
| static int DefaultFrameSizePerChannel(int sample_rate_hz) { |
| return FrameSizePerChannel(20, sample_rate_hz); |
| } |
| |
| int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst, |
| size_t channels, |
| int32_t application, |
| int sample_rate_hz) { |
| int opus_app; |
| if (!inst) |
| return -1; |
| |
| switch (application) { |
| case 0: |
| opus_app = OPUS_APPLICATION_VOIP; |
| break; |
| case 1: |
| opus_app = OPUS_APPLICATION_AUDIO; |
| break; |
| default: |
| return -1; |
| } |
| |
| OpusEncInst* state = (OpusEncInst*)calloc(1, sizeof(OpusEncInst)); |
| RTC_DCHECK(state); |
| |
| int error; |
| state->encoder = opus_encoder_create(sample_rate_hz, (int)channels, opus_app, |
| &error); |
| |
| if (error != OPUS_OK || (!state->encoder && |
| !state->multistream_encoder)) { |
| WebRtcOpus_EncoderFree(state); |
| return -1; |
| } |
| |
| state->in_dtx_mode = 0; |
| state->channels = channels; |
| |
| *inst = state; |
| return 0; |
| } |
| |
| int16_t WebRtcOpus_MultistreamEncoderCreate( |
| OpusEncInst** inst, |
| size_t channels, |
| int32_t application, |
| size_t streams, |
| size_t coupled_streams, |
| const unsigned char *channel_mapping) { |
| int opus_app; |
| if (!inst) |
| return -1; |
| |
| switch (application) { |
| case 0: |
| opus_app = OPUS_APPLICATION_VOIP; |
| break; |
| case 1: |
| opus_app = OPUS_APPLICATION_AUDIO; |
| break; |
| default: |
| return -1; |
| } |
| |
| OpusEncInst* state = (OpusEncInst*)calloc(1, sizeof(OpusEncInst)); |
| RTC_DCHECK(state); |
| |
| int error; |
| state->multistream_encoder = |
| opus_multistream_encoder_create( |
| 48000, |
| channels, |
| streams, |
| coupled_streams, |
| channel_mapping, |
| opus_app, |
| &error); |
| |
| if (error != OPUS_OK || (!state->encoder && |
| !state->multistream_encoder)) { |
| WebRtcOpus_EncoderFree(state); |
| return -1; |
| } |
| |
| state->in_dtx_mode = 0; |
| state->channels = channels; |
| |
| *inst = state; |
| return 0; |
| } |
| |
| int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) { |
| if (inst) { |
| if (inst->encoder) { |
| opus_encoder_destroy(inst->encoder); |
| } else { |
| opus_multistream_encoder_destroy(inst->multistream_encoder); |
| } |
| free(inst); |
| return 0; |
| } else { |
| return -1; |
| } |
| } |
| |
| int WebRtcOpus_Encode(OpusEncInst* inst, |
| const int16_t* audio_in, |
| size_t samples, |
| size_t length_encoded_buffer, |
| uint8_t* encoded) { |
| int res; |
| |
| if (samples > 48 * kWebRtcOpusMaxEncodeFrameSizeMs) { |
| return -1; |
| } |
| |
| if (inst->encoder) { |
| res = opus_encode(inst->encoder, |
| (const opus_int16*)audio_in, |
| (int)samples, |
| encoded, |
| (opus_int32)length_encoded_buffer); |
| } else { |
| res = opus_multistream_encode(inst->multistream_encoder, |
| (const opus_int16*)audio_in, |
| (int)samples, |
| encoded, |
| (opus_int32)length_encoded_buffer); |
| } |
| |
| if (res <= 0) { |
| return -1; |
| } |
| |
| if (res <= 2) { |
| // Indicates DTX since the packet has nothing but a header. In principle, |
| // there is no need to send this packet. However, we do transmit the first |
| // occurrence to let the decoder know that the encoder enters DTX mode. |
| if (inst->in_dtx_mode) { |
| return 0; |
| } else { |
| inst->in_dtx_mode = 1; |
| return res; |
| } |
| } |
| |
| inst->in_dtx_mode = 0; |
| return res; |
| } |
| |
| #define ENCODER_CTL(inst, vargs) ( \ |
| inst->encoder ? \ |
| opus_encoder_ctl(inst->encoder, vargs) \ |
| : opus_multistream_encoder_ctl(inst->multistream_encoder, vargs)) |
| |
| |
| int16_t WebRtcOpus_SetBitRate(OpusEncInst* inst, int32_t rate) { |
| if (inst) { |
| return ENCODER_CTL(inst, OPUS_SET_BITRATE(rate)); |
| } else { |
| return -1; |
| } |
| } |
| |
| int16_t WebRtcOpus_SetPacketLossRate(OpusEncInst* inst, int32_t loss_rate) { |
| if (inst) { |
| return ENCODER_CTL(inst, OPUS_SET_PACKET_LOSS_PERC(loss_rate)); |
| } else { |
| return -1; |
| } |
| } |
| |
| int16_t WebRtcOpus_SetMaxPlaybackRate(OpusEncInst* inst, int32_t frequency_hz) { |
| opus_int32 set_bandwidth; |
| |
| if (!inst) |
| return -1; |
| |
| if (frequency_hz <= 8000) { |
| set_bandwidth = OPUS_BANDWIDTH_NARROWBAND; |
| } else if (frequency_hz <= 12000) { |
| set_bandwidth = OPUS_BANDWIDTH_MEDIUMBAND; |
| } else if (frequency_hz <= 16000) { |
| set_bandwidth = OPUS_BANDWIDTH_WIDEBAND; |
| } else if (frequency_hz <= 24000) { |
| set_bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND; |
| } else { |
| set_bandwidth = OPUS_BANDWIDTH_FULLBAND; |
| } |
| return ENCODER_CTL(inst, OPUS_SET_MAX_BANDWIDTH(set_bandwidth)); |
| } |
| |
| int16_t WebRtcOpus_GetMaxPlaybackRate(OpusEncInst* const inst, |
| int32_t* result_hz) { |
| if (inst->encoder) { |
| if (opus_encoder_ctl( |
| inst->encoder, |
| OPUS_GET_MAX_BANDWIDTH(result_hz)) == OPUS_OK) { |
| return 0; |
| } |
| return -1; |
| } |
| |
| opus_int32 max_bandwidth; |
| int s; |
| int ret; |
| |
| max_bandwidth = 0; |
| ret = OPUS_OK; |
| s = 0; |
| while (ret == OPUS_OK) { |
| OpusEncoder *enc; |
| opus_int32 bandwidth; |
| |
| ret = ENCODER_CTL(inst, OPUS_MULTISTREAM_GET_ENCODER_STATE(s, &enc)); |
| if (ret == OPUS_BAD_ARG) |
| break; |
| if (ret != OPUS_OK) |
| return -1; |
| if (opus_encoder_ctl(enc, OPUS_GET_MAX_BANDWIDTH(&bandwidth)) != OPUS_OK) |
| return -1; |
| |
| if (max_bandwidth != 0 && max_bandwidth != bandwidth) |
| return -1; |
| |
| max_bandwidth = bandwidth; |
| s++; |
| } |
| *result_hz = max_bandwidth; |
| return 0; |
| } |
| |
| int16_t WebRtcOpus_EnableFec(OpusEncInst* inst) { |
| if (inst) { |
| return ENCODER_CTL(inst, OPUS_SET_INBAND_FEC(1)); |
| } else { |
| return -1; |
| } |
| } |
| |
| int16_t WebRtcOpus_DisableFec(OpusEncInst* inst) { |
| if (inst) { |
| return ENCODER_CTL(inst, OPUS_SET_INBAND_FEC(0)); |
| } else { |
| return -1; |
| } |
| } |
| |
| int16_t WebRtcOpus_EnableDtx(OpusEncInst* inst) { |
| if (!inst) { |
| return -1; |
| } |
| |
| // To prevent Opus from entering CELT-only mode by forcing signal type to |
| // voice to make sure that DTX behaves correctly. Currently, DTX does not |
| // last long during a pure silence, if the signal type is not forced. |
| // TODO(minyue): Remove the signal type forcing when Opus DTX works properly |
| // without it. |
| int ret = ENCODER_CTL(inst, |
| OPUS_SET_SIGNAL(OPUS_SIGNAL_VOICE)); |
| if (ret != OPUS_OK) |
| return ret; |
| |
| return ENCODER_CTL(inst, OPUS_SET_DTX(1)); |
| } |
| |
| int16_t WebRtcOpus_DisableDtx(OpusEncInst* inst) { |
| if (inst) { |
| int ret = ENCODER_CTL(inst, |
| OPUS_SET_SIGNAL(OPUS_AUTO)); |
| if (ret != OPUS_OK) |
| return ret; |
| return ENCODER_CTL(inst, OPUS_SET_DTX(0)); |
| } else { |
| return -1; |
| } |
| } |
| |
| int16_t WebRtcOpus_EnableCbr(OpusEncInst* inst) { |
| if (inst) { |
| return ENCODER_CTL(inst, OPUS_SET_VBR(0)); |
| } else { |
| return -1; |
| } |
| } |
| |
| int16_t WebRtcOpus_DisableCbr(OpusEncInst* inst) { |
| if (inst) { |
| return ENCODER_CTL(inst, OPUS_SET_VBR(1)); |
| } else { |
| return -1; |
| } |
| } |
| |
| int16_t WebRtcOpus_SetComplexity(OpusEncInst* inst, int32_t complexity) { |
| if (inst) { |
| return ENCODER_CTL(inst, |
| OPUS_SET_COMPLEXITY(complexity)); |
| } else { |
| return -1; |
| } |
| } |
| |
| int32_t WebRtcOpus_GetBandwidth(OpusEncInst* inst) { |
| if (!inst) { |
| return -1; |
| } |
| int32_t bandwidth; |
| if (ENCODER_CTL(inst, |
| OPUS_GET_BANDWIDTH(&bandwidth)) == 0) { |
| return bandwidth; |
| } else { |
| return -1; |
| } |
| |
| } |
| |
| int16_t WebRtcOpus_SetBandwidth(OpusEncInst* inst, int32_t bandwidth) { |
| if (inst) { |
| return ENCODER_CTL(inst, |
| OPUS_SET_BANDWIDTH(bandwidth)); |
| } else { |
| return -1; |
| } |
| } |
| |
| int16_t WebRtcOpus_SetForceChannels(OpusEncInst* inst, size_t num_channels) { |
| if (!inst) |
| return -1; |
| if (num_channels == 0) { |
| return ENCODER_CTL(inst, |
| OPUS_SET_FORCE_CHANNELS(OPUS_AUTO)); |
| } else if (num_channels == 1 || num_channels == 2) { |
| return ENCODER_CTL(inst, |
| OPUS_SET_FORCE_CHANNELS(num_channels)); |
| } else { |
| return -1; |
| } |
| } |
| |
| int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, |
| size_t channels, |
| int sample_rate_hz) { |
| int error; |
| OpusDecInst* state; |
| |
| if (inst != NULL) { |
| // Create Opus decoder state. |
| state = (OpusDecInst*) calloc(1, sizeof(OpusDecInst)); |
| if (state == NULL) { |
| return -1; |
| } |
| |
| state->decoder = opus_decoder_create(sample_rate_hz, (int)channels, &error); |
| if (error == OPUS_OK && state->decoder) { |
| // Creation of memory all ok. |
| state->channels = channels; |
| state->prev_decoded_samples = DefaultFrameSizePerChannel(sample_rate_hz); |
| state->in_dtx_mode = 0; |
| state->sample_rate_hz = sample_rate_hz; |
| *inst = state; |
| return 0; |
| } |
| |
| // If memory allocation was unsuccessful, free the entire state. |
| if (state->decoder) { |
| opus_decoder_destroy(state->decoder); |
| } |
| free(state); |
| } |
| return -1; |
| } |
| |
| int16_t WebRtcOpus_MultistreamDecoderCreate( |
| OpusDecInst** inst, size_t channels, |
| size_t streams, |
| size_t coupled_streams, |
| const unsigned char* channel_mapping) { |
| int error; |
| OpusDecInst* state; |
| |
| if (inst != NULL) { |
| // Create Opus decoder state. |
| state = (OpusDecInst*) calloc(1, sizeof(OpusDecInst)); |
| if (state == NULL) { |
| return -1; |
| } |
| |
| // Create new memory, always at 48000 Hz. |
| state->multistream_decoder = opus_multistream_decoder_create( |
| 48000, channels, |
| streams, |
| coupled_streams, |
| channel_mapping, |
| &error); |
| |
| if (error == OPUS_OK && state->multistream_decoder) { |
| // Creation of memory all ok. |
| state->channels = channels; |
| state->prev_decoded_samples = DefaultFrameSizePerChannel(48000); |
| state->in_dtx_mode = 0; |
| state->sample_rate_hz = 48000; |
| *inst = state; |
| return 0; |
| } |
| |
| // If memory allocation was unsuccessful, free the entire state. |
| opus_multistream_decoder_destroy(state->multistream_decoder); |
| free(state); |
| } |
| return -1; |
| } |
| |
| int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst) { |
| if (inst) { |
| if (inst->decoder) { |
| opus_decoder_destroy(inst->decoder); |
| } else if (inst->multistream_decoder) { |
| opus_multistream_decoder_destroy(inst->multistream_decoder); |
| } |
| free(inst); |
| return 0; |
| } else { |
| return -1; |
| } |
| } |
| |
| size_t WebRtcOpus_DecoderChannels(OpusDecInst* inst) { |
| return inst->channels; |
| } |
| |
| void WebRtcOpus_DecoderInit(OpusDecInst* inst) { |
| if (inst->decoder) { |
| opus_decoder_ctl(inst->decoder, OPUS_RESET_STATE); |
| } else { |
| opus_multistream_decoder_ctl(inst->multistream_decoder, |
| OPUS_RESET_STATE); |
| } |
| inst->in_dtx_mode = 0; |
| } |
| |
| /* For decoder to determine if it is to output speech or comfort noise. */ |
| static int16_t DetermineAudioType(OpusDecInst* inst, size_t encoded_bytes) { |
| // Audio type becomes comfort noise if |encoded_byte| is 1 and keeps |
| // to be so if the following |encoded_byte| are 0 or 1. |
| if (encoded_bytes == 0 && inst->in_dtx_mode) { |
| return 2; // Comfort noise. |
| } else if (encoded_bytes == 1 || encoded_bytes == 2) { |
| // TODO(henrik.lundin): There is a slight risk that a 2-byte payload is in |
| // fact a 1-byte TOC with a 1-byte payload. That will be erroneously |
| // interpreted as comfort noise output, but such a payload is probably |
| // faulty anyway. |
| |
| // TODO(webrtc:10218): This is wrong for multistream opus. Then are several |
| // single-stream packets glued together with some packet size bytes in |
| // between. See https://tools.ietf.org/html/rfc6716#appendix-B |
| inst->in_dtx_mode = 1; |
| return 2; // Comfort noise. |
| } else { |
| inst->in_dtx_mode = 0; |
| return 0; // Speech. |
| } |
| } |
| |
| /* |frame_size| is set to maximum Opus frame size in the normal case, and |
| * is set to the number of samples needed for PLC in case of losses. |
| * It is up to the caller to make sure the value is correct. */ |
| static int DecodeNative(OpusDecInst* inst, const uint8_t* encoded, |
| size_t encoded_bytes, int frame_size, |
| int16_t* decoded, int16_t* audio_type, int decode_fec) { |
| int res = -1; |
| if (inst->decoder) { |
| res = opus_decode(inst->decoder, encoded, (opus_int32)encoded_bytes, |
| (opus_int16*)decoded, frame_size, decode_fec); |
| } else { |
| res = opus_multistream_decode( |
| inst->multistream_decoder, encoded, (opus_int32)encoded_bytes, |
| (opus_int16*)decoded, frame_size, decode_fec); |
| } |
| |
| if (res <= 0) |
| return -1; |
| |
| *audio_type = DetermineAudioType(inst, encoded_bytes); |
| |
| return res; |
| } |
| |
| int WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded, |
| size_t encoded_bytes, int16_t* decoded, |
| int16_t* audio_type) { |
| int decoded_samples; |
| |
| if (encoded_bytes == 0) { |
| *audio_type = DetermineAudioType(inst, encoded_bytes); |
| decoded_samples = WebRtcOpus_DecodePlc(inst, decoded, 1); |
| } else { |
| decoded_samples = DecodeNative(inst, encoded, encoded_bytes, |
| MaxFrameSizePerChannel(inst->sample_rate_hz), |
| decoded, audio_type, 0); |
| } |
| if (decoded_samples < 0) { |
| return -1; |
| } |
| |
| /* Update decoded sample memory, to be used by the PLC in case of losses. */ |
| inst->prev_decoded_samples = decoded_samples; |
| |
| return decoded_samples; |
| } |
| |
| int WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded, |
| int number_of_lost_frames) { |
| int16_t audio_type = 0; |
| int decoded_samples; |
| int plc_samples; |
| |
| /* The number of samples we ask for is |number_of_lost_frames| times |
| * |prev_decoded_samples_|. Limit the number of samples to maximum |
| * |MaxFrameSizePerChannel()|. */ |
| plc_samples = number_of_lost_frames * inst->prev_decoded_samples; |
| const int max_samples_per_channel = |
| MaxFrameSizePerChannel(inst->sample_rate_hz); |
| plc_samples = plc_samples <= max_samples_per_channel |
| ? plc_samples |
| : max_samples_per_channel; |
| decoded_samples = DecodeNative(inst, NULL, 0, plc_samples, |
| decoded, &audio_type, 0); |
| if (decoded_samples < 0) { |
| return -1; |
| } |
| |
| return decoded_samples; |
| } |
| |
| int WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded, |
| size_t encoded_bytes, int16_t* decoded, |
| int16_t* audio_type) { |
| int decoded_samples; |
| int fec_samples; |
| |
| if (WebRtcOpus_PacketHasFec(encoded, encoded_bytes) != 1) { |
| return 0; |
| } |
| |
| fec_samples = |
| opus_packet_get_samples_per_frame(encoded, inst->sample_rate_hz); |
| |
| decoded_samples = DecodeNative(inst, encoded, encoded_bytes, |
| fec_samples, decoded, audio_type, 1); |
| if (decoded_samples < 0) { |
| return -1; |
| } |
| |
| return decoded_samples; |
| } |
| |
| int WebRtcOpus_DurationEst(OpusDecInst* inst, |
| const uint8_t* payload, |
| size_t payload_length_bytes) { |
| if (payload_length_bytes == 0) { |
| // WebRtcOpus_Decode calls PLC when payload length is zero. So we return |
| // PLC duration correspondingly. |
| return WebRtcOpus_PlcDuration(inst); |
| } |
| |
| int frames, samples; |
| frames = opus_packet_get_nb_frames(payload, (opus_int32)payload_length_bytes); |
| if (frames < 0) { |
| /* Invalid payload data. */ |
| return 0; |
| } |
| samples = |
| frames * opus_packet_get_samples_per_frame(payload, inst->sample_rate_hz); |
| if (samples > 120 * inst->sample_rate_hz / 1000) { |
| // More than 120 ms' worth of samples. |
| return 0; |
| } |
| return samples; |
| } |
| |
| int WebRtcOpus_PlcDuration(OpusDecInst* inst) { |
| /* The number of samples we ask for is |number_of_lost_frames| times |
| * |prev_decoded_samples_|. Limit the number of samples to maximum |
| * |MaxFrameSizePerChannel()|. */ |
| const int plc_samples = inst->prev_decoded_samples; |
| const int max_samples_per_channel = |
| MaxFrameSizePerChannel(inst->sample_rate_hz); |
| return plc_samples <= max_samples_per_channel ? plc_samples |
| : max_samples_per_channel; |
| } |
| |
| int WebRtcOpus_FecDurationEst(const uint8_t* payload, |
| size_t payload_length_bytes, |
| int sample_rate_hz) { |
| if (WebRtcOpus_PacketHasFec(payload, payload_length_bytes) != 1) { |
| return 0; |
| } |
| const int samples = |
| opus_packet_get_samples_per_frame(payload, sample_rate_hz); |
| const int samples_per_ms = sample_rate_hz / 1000; |
| if (samples < 10 * samples_per_ms || samples > 120 * samples_per_ms) { |
| /* Invalid payload duration. */ |
| return 0; |
| } |
| return samples; |
| } |
| |
| int WebRtcOpus_PacketHasFec(const uint8_t* payload, |
| size_t payload_length_bytes) { |
| int frames, channels, payload_length_ms; |
| int n; |
| opus_int16 frame_sizes[48]; |
| const unsigned char *frame_data[48]; |
| |
| if (payload == NULL || payload_length_bytes == 0) |
| return 0; |
| |
| /* In CELT_ONLY mode, packets should not have FEC. */ |
| if (payload[0] & 0x80) |
| return 0; |
| |
| // For computing the payload length in ms, the sample rate is not important |
| // since it cancels out. We use 48 kHz, but any valid sample rate would work. |
| payload_length_ms = opus_packet_get_samples_per_frame(payload, 48000) / 48; |
| if (10 > payload_length_ms) |
| payload_length_ms = 10; |
| |
| channels = opus_packet_get_nb_channels(payload); |
| |
| switch (payload_length_ms) { |
| case 10: |
| case 20: { |
| frames = 1; |
| break; |
| } |
| case 40: { |
| frames = 2; |
| break; |
| } |
| case 60: { |
| frames = 3; |
| break; |
| } |
| default: { |
| return 0; // It is actually even an invalid packet. |
| } |
| } |
| |
| /* The following is to parse the LBRR flags. */ |
| if (opus_packet_parse(payload, (opus_int32)payload_length_bytes, NULL, |
| frame_data, frame_sizes, NULL) < 0) { |
| return 0; |
| } |
| |
| if (frame_sizes[0] <= 1) { |
| return 0; |
| } |
| |
| for (n = 0; n < channels; n++) { |
| if (frame_data[0][0] & (0x80 >> ((n + 1) * (frames + 1) - 1))) |
| return 1; |
| } |
| |
| return 0; |
| } |
