diff options
Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_interface.cc')
| -rw-r--r-- | third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_interface.cc | 880 |
1 files changed, 880 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_interface.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_interface.cc new file mode 100644 index 0000000000..64a1f59237 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_interface.cc @@ -0,0 +1,880 @@ +/* + * 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; +} |