| /* |
| * 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 <cstdlib> |
| |
| #include <numeric> |
| |
| #include "api/array_view.h" |
| #include "rtc_base/checks.h" |
| #include "system_wrappers/include/field_trial.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, |
| |
| // Duration of audio that each call to packet loss concealment covers. |
| kWebRtcOpusPlcFrameSizeMs = 10, |
| }; |
| |
| constexpr char kPlcUsePrevDecodedSamplesFieldTrial[] = |
| "WebRTC-Audio-OpusPlcUsePrevDecodedSamples"; |
| |
| constexpr char kAvoidNoisePumpingDuringDtxFieldTrial[] = |
| "WebRTC-Audio-OpusAvoidNoisePumpingDuringDtx"; |
| |
| constexpr char kSetSignalVoiceWithDtxFieldTrial[] = |
| "WebRTC-Audio-OpusSetSignalVoiceWithDtx"; |
| |
| 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); |
| } |
| |
| // Returns true if the `encoded` payload corresponds to a refresh DTX packet |
| // whose energy is larger than the expected for non activity packets. |
| static bool WebRtcOpus_IsHighEnergyRefreshDtxPacket( |
| OpusEncInst* inst, |
| rtc::ArrayView<const int16_t> frame, |
| rtc::ArrayView<const uint8_t> encoded) { |
| if (encoded.size() <= 2) { |
| return false; |
| } |
| int number_frames = |
| frame.size() / DefaultFrameSizePerChannel(inst->sample_rate_hz); |
| if (number_frames > 0 && |
| WebRtcOpus_PacketHasVoiceActivity(encoded.data(), encoded.size()) == 0) { |
| const float average_frame_energy = |
| std::accumulate(frame.begin(), frame.end(), 0.0f, |
| [](float a, int32_t b) { return a + b * b; }) / |
| number_frames; |
| if (WebRtcOpus_GetInDtx(inst) == 1 && |
| average_frame_energy >= inst->smooth_energy_non_active_frames * 0.5f) { |
| // This is a refresh DTX packet as the encoder is in DTX and has |
| // produced a payload > 2 bytes. This refresh packet has a higher energy |
| // than the smooth energy of non activity frames (with a 3 dB negative |
| // margin) and, therefore, it is flagged as a high energy refresh DTX |
| // packet. |
| return true; |
| } |
| // The average energy is tracked in a similar way as the modeling of the |
| // comfort noise in the Silk decoder in Opus |
| // (third_party/opus/src/silk/CNG.c). |
| if (average_frame_energy < inst->smooth_energy_non_active_frames * 0.5f) { |
| inst->smooth_energy_non_active_frames = average_frame_energy; |
| } else { |
| inst->smooth_energy_non_active_frames += |
| (average_frame_energy - inst->smooth_energy_non_active_frames) * |
| 0.25f; |
| } |
| } |
| return false; |
| } |
| |
| 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 = |
| reinterpret_cast<OpusEncInst*>(calloc(1, sizeof(OpusEncInst))); |
| RTC_DCHECK(state); |
| |
| int error; |
| state->encoder = opus_encoder_create( |
| sample_rate_hz, static_cast<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; |
| state->sample_rate_hz = sample_rate_hz; |
| state->smooth_energy_non_active_frames = 0.0f; |
| state->avoid_noise_pumping_during_dtx = |
| webrtc::field_trial::IsEnabled(kAvoidNoisePumpingDuringDtxFieldTrial); |
| |
| *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 = |
| reinterpret_cast<OpusEncInst*>(calloc(1, sizeof(OpusEncInst))); |
| RTC_DCHECK(state); |
| |
| int error; |
| const int sample_rate_hz = 48000; |
| state->multistream_encoder = opus_multistream_encoder_create( |
| sample_rate_hz, 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; |
| state->sample_rate_hz = sample_rate_hz; |
| state->smooth_energy_non_active_frames = 0.0f; |
| state->avoid_noise_pumping_during_dtx = false; |
| |
| *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, |
| static_cast<int>(samples), encoded, |
| static_cast<opus_int32>(length_encoded_buffer)); |
| } else { |
| res = opus_multistream_encode( |
| inst->multistream_encoder, (const opus_int16*)audio_in, |
| static_cast<int>(samples), encoded, |
| static_cast<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; |
| } |
| } |
| |
| if (inst->avoid_noise_pumping_during_dtx && WebRtcOpus_GetUseDtx(inst) == 1 && |
| WebRtcOpus_IsHighEnergyRefreshDtxPacket( |
| inst, rtc::MakeArrayView(audio_in, samples), |
| rtc::MakeArrayView(encoded, res))) { |
| // This packet is a high energy refresh DTX packet. For avoiding an increase |
| // of the energy in the DTX region at the decoder, this packet is |
| // substituted by a TOC byte with one empty frame. |
| // The number of frames described in the TOC byte |
| // (https://tools.ietf.org/html/rfc6716#section-3.1) are overwritten to |
| // always indicate one frame (last two bits equal to 0). |
| encoded[0] = encoded[0] & 0b11111100; |
| inst->in_dtx_mode = 1; |
| // The payload is just the TOC byte and has 1 byte as length. |
| return 1; |
| } |
| 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) { |
| if (webrtc::field_trial::IsEnabled(kSetSignalVoiceWithDtxFieldTrial)) { |
| 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)); |
| } else { |
| return -1; |
| } |
| } |
| |
| int16_t WebRtcOpus_DisableDtx(OpusEncInst* inst) { |
| if (inst) { |
| if (webrtc::field_trial::IsEnabled(kSetSignalVoiceWithDtxFieldTrial)) { |
| 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_GetUseDtx(OpusEncInst* inst) { |
| if (inst) { |
| opus_int32 use_dtx; |
| if (ENCODER_CTL(inst, OPUS_GET_DTX(&use_dtx)) == 0) { |
| return use_dtx; |
| } |
| } |
| 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; |
| } |
| } |
| |
| int32_t WebRtcOpus_GetInDtx(OpusEncInst* inst) { |
| if (!inst) { |
| return -1; |
| } |
| #ifdef OPUS_GET_IN_DTX |
| int32_t in_dtx; |
| if (ENCODER_CTL(inst, OPUS_GET_IN_DTX(&in_dtx)) == 0) { |
| return in_dtx; |
| } |
| #endif |
| 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 = reinterpret_cast<OpusDecInst*>(calloc(1, sizeof(OpusDecInst))); |
| if (state == NULL) { |
| return -1; |
| } |
| |
| state->decoder = |
| opus_decoder_create(sample_rate_hz, static_cast<int>(channels), &error); |
| if (error == OPUS_OK && state->decoder) { |
| // Creation of memory all ok. |
| state->channels = channels; |
| state->sample_rate_hz = sample_rate_hz; |
| state->plc_use_prev_decoded_samples = |
| webrtc::field_trial::IsEnabled(kPlcUsePrevDecodedSamplesFieldTrial); |
| if (state->plc_use_prev_decoded_samples) { |
| state->prev_decoded_samples = |
| DefaultFrameSizePerChannel(state->sample_rate_hz); |
| } |
| state->in_dtx_mode = 0; |
| *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 = reinterpret_cast<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->sample_rate_hz = 48000; |
| state->plc_use_prev_decoded_samples = |
| webrtc::field_trial::IsEnabled(kPlcUsePrevDecodedSamplesFieldTrial); |
| if (state->plc_use_prev_decoded_samples) { |
| state->prev_decoded_samples = |
| DefaultFrameSizePerChannel(state->sample_rate_hz); |
| } |
| state->in_dtx_mode = 0; |
| *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, static_cast<opus_int32>(encoded_bytes), |
| reinterpret_cast<opus_int16*>(decoded), frame_size, decode_fec); |
| } else { |
| res = opus_multistream_decode(inst->multistream_decoder, encoded, |
| static_cast<opus_int32>(encoded_bytes), |
| reinterpret_cast<opus_int16*>(decoded), |
| frame_size, decode_fec); |
| } |
| |
| if (res <= 0) |
| return -1; |
| |
| *audio_type = DetermineAudioType(inst, encoded_bytes); |
| |
| return res; |
| } |
| |
| static int DecodePlc(OpusDecInst* inst, int16_t* decoded) { |
| int16_t audio_type = 0; |
| int decoded_samples; |
| int plc_samples = |
| FrameSizePerChannel(kWebRtcOpusPlcFrameSizeMs, inst->sample_rate_hz); |
| |
| if (inst->plc_use_prev_decoded_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 = 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_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 = DecodePlc(inst, decoded); |
| } else { |
| decoded_samples = DecodeNative(inst, encoded, encoded_bytes, |
| MaxFrameSizePerChannel(inst->sample_rate_hz), |
| decoded, audio_type, 0); |
| } |
| if (decoded_samples < 0) { |
| return -1; |
| } |
| |
| if (inst->plc_use_prev_decoded_samples) { |
| /* 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_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, static_cast<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) { |
| if (inst->plc_use_prev_decoded_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()`. */ |
| 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; |
| } |
| return FrameSizePerChannel(kWebRtcOpusPlcFrameSizeMs, inst->sample_rate_hz); |
| } |
| |
| 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_NumSilkFrames(const uint8_t* payload) { |
| // 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. |
| int payload_length_ms = |
| opus_packet_get_samples_per_frame(payload, 48000) / 48; |
| if (payload_length_ms < 10) |
| payload_length_ms = 10; |
| |
| int silk_frames; |
| switch (payload_length_ms) { |
| case 10: |
| case 20: |
| silk_frames = 1; |
| break; |
| case 40: |
| silk_frames = 2; |
| break; |
| case 60: |
| silk_frames = 3; |
| break; |
| default: |
| return 0; // It is actually even an invalid packet. |
| } |
| return silk_frames; |
| } |
| |
| // This method is based on Definition of the Opus Audio Codec |
| // (https://tools.ietf.org/html/rfc6716). Basically, this method is based on |
| // parsing the LP layer of an Opus packet, particularly the LBRR flag. |
| int WebRtcOpus_PacketHasFec(const uint8_t* payload, |
| size_t payload_length_bytes) { |
| if (payload == NULL || payload_length_bytes == 0) |
| return 0; |
| |
| // In CELT_ONLY mode, packets should not have FEC. |
| if (payload[0] & 0x80) |
| return 0; |
| |
| int silk_frames = WebRtcOpus_NumSilkFrames(payload); |
| if (silk_frames == 0) |
| return 0; // Not valid. |
| |
| const int channels = opus_packet_get_nb_channels(payload); |
| RTC_DCHECK(channels == 1 || channels == 2); |
| |
| // Max number of frames in an Opus packet is 48. |
| opus_int16 frame_sizes[48]; |
| const unsigned char* frame_data[48]; |
| |
| // Parse packet to get the frames. But we only care about the first frame, |
| // since we can only decode the FEC from the first one. |
| if (opus_packet_parse(payload, static_cast<opus_int32>(payload_length_bytes), |
| NULL, frame_data, frame_sizes, NULL) < 0) { |
| return 0; |
| } |
| |
| if (frame_sizes[0] < 1) { |
| return 0; |
| } |
| |
| // A frame starts with the LP layer. The LP layer begins with two to eight |
| // header bits.These consist of one VAD bit per SILK frame (up to 3), |
| // followed by a single flag indicating the presence of LBRR frames. |
| // For a stereo packet, these first flags correspond to the mid channel, and |
| // a second set of flags is included for the side channel. Because these are |
| // the first symbols decoded by the range coder and because they are coded |
| // as binary values with uniform probability, they can be extracted directly |
| // from the most significant bits of the first byte of compressed data. |
| for (int n = 0; n < channels; n++) { |
| // The LBRR bit for channel 1 is on the (`silk_frames` + 1)-th bit, and |
| // that of channel 2 is on the |(`silk_frames` + 1) * 2 + 1|-th bit. |
| if (frame_data[0][0] & (0x80 >> ((n + 1) * (silk_frames + 1) - 1))) |
| return 1; |
| } |
| |
| return 0; |
| } |
| |
| int WebRtcOpus_PacketHasVoiceActivity(const uint8_t* payload, |
| size_t payload_length_bytes) { |
| if (payload == NULL || payload_length_bytes == 0) |
| return 0; |
| |
| // In CELT_ONLY mode we can not determine whether there is VAD. |
| if (payload[0] & 0x80) |
| return -1; |
| |
| int silk_frames = WebRtcOpus_NumSilkFrames(payload); |
| if (silk_frames == 0) |
| return -1; |
| |
| const int channels = opus_packet_get_nb_channels(payload); |
| RTC_DCHECK(channels == 1 || channels == 2); |
| |
| // Max number of frames in an Opus packet is 48. |
| opus_int16 frame_sizes[48]; |
| const unsigned char* frame_data[48]; |
| |
| // Parse packet to get the frames. |
| int frames = |
| opus_packet_parse(payload, static_cast<opus_int32>(payload_length_bytes), |
| NULL, frame_data, frame_sizes, NULL); |
| if (frames < 0) |
| return -1; |
| |
| // Iterate over all Opus frames which may contain multiple SILK frames. |
| for (int frame = 0; frame < frames; frame++) { |
| if (frame_sizes[frame] < 1) { |
| continue; |
| } |
| if (frame_data[frame][0] >> (8 - silk_frames)) |
| return 1; |
| if (channels == 2 && |
| (frame_data[frame][0] << (silk_frames + 1)) >> (8 - silk_frames)) |
| return 1; |
| } |
| |
| return 0; |
| } |
