| /* |
| * Copyright (c) 2014 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_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_ |
| #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_ |
| |
| #include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h" |
| |
| #include <algorithm> |
| |
| #include "webrtc/base/checks.h" |
| #include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h" |
| #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| |
| namespace webrtc { |
| |
| const int kIsacPayloadType = 103; |
| const int kDefaultBitRate = 32000; |
| |
| template <typename T> |
| AudioEncoderDecoderIsacT<T>::Config::Config() |
| : payload_type(kIsacPayloadType), |
| sample_rate_hz(16000), |
| frame_size_ms(30), |
| bit_rate(kDefaultBitRate), |
| max_bit_rate(-1), |
| max_payload_size_bytes(-1) { |
| } |
| |
| template <typename T> |
| bool AudioEncoderDecoderIsacT<T>::Config::IsOk() const { |
| if (max_bit_rate < 32000 && max_bit_rate != -1) |
| return false; |
| if (max_payload_size_bytes < 120 && max_payload_size_bytes != -1) |
| return false; |
| switch (sample_rate_hz) { |
| case 16000: |
| if (max_bit_rate > 53400) |
| return false; |
| if (max_payload_size_bytes > 400) |
| return false; |
| return (frame_size_ms == 30 || frame_size_ms == 60) && |
| ((bit_rate >= 10000 && bit_rate <= 32000) || bit_rate == 0); |
| case 32000: |
| case 48000: |
| if (max_bit_rate > 160000) |
| return false; |
| if (max_payload_size_bytes > 600) |
| return false; |
| return T::has_swb && |
| (frame_size_ms == 30 && |
| ((bit_rate >= 10000 && bit_rate <= 56000) || bit_rate == 0)); |
| default: |
| return false; |
| } |
| } |
| |
| template <typename T> |
| AudioEncoderDecoderIsacT<T>::ConfigAdaptive::ConfigAdaptive() |
| : payload_type(kIsacPayloadType), |
| sample_rate_hz(16000), |
| initial_frame_size_ms(30), |
| initial_bit_rate(kDefaultBitRate), |
| max_bit_rate(-1), |
| enforce_frame_size(false), |
| max_payload_size_bytes(-1) { |
| } |
| |
| template <typename T> |
| bool AudioEncoderDecoderIsacT<T>::ConfigAdaptive::IsOk() const { |
| if (max_bit_rate < 32000 && max_bit_rate != -1) |
| return false; |
| if (max_payload_size_bytes < 120 && max_payload_size_bytes != -1) |
| return false; |
| switch (sample_rate_hz) { |
| case 16000: |
| if (max_bit_rate > 53400) |
| return false; |
| if (max_payload_size_bytes > 400) |
| return false; |
| return (initial_frame_size_ms == 30 || initial_frame_size_ms == 60) && |
| initial_bit_rate >= 10000 && initial_bit_rate <= 32000; |
| case 32000: |
| case 48000: |
| if (max_bit_rate > 160000) |
| return false; |
| if (max_payload_size_bytes > 600) |
| return false; |
| return T::has_swb && |
| (initial_frame_size_ms == 30 && initial_bit_rate >= 10000 && |
| initial_bit_rate <= 56000); |
| default: |
| return false; |
| } |
| } |
| |
| template <typename T> |
| AudioEncoderDecoderIsacT<T>::AudioEncoderDecoderIsacT(const Config& config) |
| : payload_type_(config.payload_type), |
| state_lock_(CriticalSectionWrapper::CreateCriticalSection()), |
| decoder_sample_rate_hz_(0), |
| lock_(CriticalSectionWrapper::CreateCriticalSection()), |
| packet_in_progress_(false) { |
| CHECK(config.IsOk()); |
| CHECK_EQ(0, T::Create(&isac_state_)); |
| CHECK_EQ(0, T::EncoderInit(isac_state_, 1)); |
| CHECK_EQ(0, T::SetEncSampRate(isac_state_, config.sample_rate_hz)); |
| CHECK_EQ(0, T::Control(isac_state_, config.bit_rate == 0 ? kDefaultBitRate |
| : config.bit_rate, |
| config.frame_size_ms)); |
| // When config.sample_rate_hz is set to 48000 Hz (iSAC-fb), the decoder is |
| // still set to 32000 Hz, since there is no full-band mode in the decoder. |
| CHECK_EQ(0, T::SetDecSampRate(isac_state_, |
| std::min(config.sample_rate_hz, 32000))); |
| if (config.max_payload_size_bytes != -1) |
| CHECK_EQ(0, |
| T::SetMaxPayloadSize(isac_state_, config.max_payload_size_bytes)); |
| if (config.max_bit_rate != -1) |
| CHECK_EQ(0, T::SetMaxRate(isac_state_, config.max_bit_rate)); |
| } |
| |
| template <typename T> |
| AudioEncoderDecoderIsacT<T>::AudioEncoderDecoderIsacT( |
| const ConfigAdaptive& config) |
| : payload_type_(config.payload_type), |
| state_lock_(CriticalSectionWrapper::CreateCriticalSection()), |
| decoder_sample_rate_hz_(0), |
| lock_(CriticalSectionWrapper::CreateCriticalSection()), |
| packet_in_progress_(false) { |
| CHECK(config.IsOk()); |
| CHECK_EQ(0, T::Create(&isac_state_)); |
| CHECK_EQ(0, T::EncoderInit(isac_state_, 0)); |
| CHECK_EQ(0, T::SetEncSampRate(isac_state_, config.sample_rate_hz)); |
| CHECK_EQ(0, T::ControlBwe(isac_state_, config.initial_bit_rate, |
| config.initial_frame_size_ms, |
| config.enforce_frame_size)); |
| CHECK_EQ(0, T::SetDecSampRate(isac_state_, config.sample_rate_hz)); |
| if (config.max_payload_size_bytes != -1) |
| CHECK_EQ(0, |
| T::SetMaxPayloadSize(isac_state_, config.max_payload_size_bytes)); |
| if (config.max_bit_rate != -1) |
| CHECK_EQ(0, T::SetMaxRate(isac_state_, config.max_bit_rate)); |
| } |
| |
| template <typename T> |
| AudioEncoderDecoderIsacT<T>::~AudioEncoderDecoderIsacT() { |
| CHECK_EQ(0, T::Free(isac_state_)); |
| } |
| |
| template <typename T> |
| int AudioEncoderDecoderIsacT<T>::SampleRateHz() const { |
| CriticalSectionScoped cs(state_lock_.get()); |
| return T::EncSampRate(isac_state_); |
| } |
| |
| template <typename T> |
| int AudioEncoderDecoderIsacT<T>::NumChannels() const { |
| return 1; |
| } |
| |
| template <typename T> |
| size_t AudioEncoderDecoderIsacT<T>::MaxEncodedBytes() const { |
| return kSufficientEncodeBufferSizeBytes; |
| } |
| |
| template <typename T> |
| int AudioEncoderDecoderIsacT<T>::Num10MsFramesInNextPacket() const { |
| CriticalSectionScoped cs(state_lock_.get()); |
| const int samples_in_next_packet = T::GetNewFrameLen(isac_state_); |
| return rtc::CheckedDivExact(samples_in_next_packet, |
| rtc::CheckedDivExact(SampleRateHz(), 100)); |
| } |
| |
| template <typename T> |
| int AudioEncoderDecoderIsacT<T>::Max10MsFramesInAPacket() const { |
| return 6; // iSAC puts at most 60 ms in a packet. |
| } |
| |
| template <typename T> |
| void AudioEncoderDecoderIsacT<T>::EncodeInternal(uint32_t rtp_timestamp, |
| const int16_t* audio, |
| size_t max_encoded_bytes, |
| uint8_t* encoded, |
| EncodedInfo* info) { |
| CriticalSectionScoped cs_lock(lock_.get()); |
| if (!packet_in_progress_) { |
| // Starting a new packet; remember the timestamp for later. |
| packet_in_progress_ = true; |
| packet_timestamp_ = rtp_timestamp; |
| } |
| int r; |
| { |
| CriticalSectionScoped cs(state_lock_.get()); |
| r = T::Encode(isac_state_, audio, encoded); |
| } |
| CHECK_GE(r, 0); |
| |
| // T::Encode doesn't allow us to tell it the size of the output |
| // buffer. All we can do is check for an overrun after the fact. |
| CHECK(static_cast<size_t>(r) <= max_encoded_bytes); |
| |
| info->encoded_bytes = r; |
| if (r == 0) |
| return; |
| |
| // Got enough input to produce a packet. Return the saved timestamp from |
| // the first chunk of input that went into the packet. |
| packet_in_progress_ = false; |
| info->encoded_timestamp = packet_timestamp_; |
| info->payload_type = payload_type_; |
| } |
| |
| template <typename T> |
| int AudioEncoderDecoderIsacT<T>::DecodeInternal(const uint8_t* encoded, |
| size_t encoded_len, |
| int sample_rate_hz, |
| int16_t* decoded, |
| SpeechType* speech_type) { |
| CriticalSectionScoped cs(state_lock_.get()); |
| CHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000) |
| << "Unsupported sample rate " << sample_rate_hz; |
| if (sample_rate_hz != decoder_sample_rate_hz_) { |
| CHECK_EQ(0, T::SetDecSampRate(isac_state_, sample_rate_hz)); |
| decoder_sample_rate_hz_ = sample_rate_hz; |
| } |
| int16_t temp_type = 1; // Default is speech. |
| int16_t ret = |
| T::DecodeInternal(isac_state_, encoded, static_cast<int16_t>(encoded_len), |
| decoded, &temp_type); |
| *speech_type = ConvertSpeechType(temp_type); |
| return ret; |
| } |
| |
| template <typename T> |
| bool AudioEncoderDecoderIsacT<T>::HasDecodePlc() const { |
| return true; |
| } |
| |
| template <typename T> |
| int AudioEncoderDecoderIsacT<T>::DecodePlc(int num_frames, int16_t* decoded) { |
| CriticalSectionScoped cs(state_lock_.get()); |
| return T::DecodePlc(isac_state_, decoded, num_frames); |
| } |
| |
| template <typename T> |
| int AudioEncoderDecoderIsacT<T>::Init() { |
| CriticalSectionScoped cs(state_lock_.get()); |
| return T::DecoderInit(isac_state_); |
| } |
| |
| template <typename T> |
| int AudioEncoderDecoderIsacT<T>::IncomingPacket(const uint8_t* payload, |
| size_t payload_len, |
| uint16_t rtp_sequence_number, |
| uint32_t rtp_timestamp, |
| uint32_t arrival_timestamp) { |
| CriticalSectionScoped cs(state_lock_.get()); |
| return T::UpdateBwEstimate( |
| isac_state_, payload, static_cast<int32_t>(payload_len), |
| rtp_sequence_number, rtp_timestamp, arrival_timestamp); |
| } |
| |
| template <typename T> |
| int AudioEncoderDecoderIsacT<T>::ErrorCode() { |
| CriticalSectionScoped cs(state_lock_.get()); |
| return T::GetErrorCode(isac_state_); |
| } |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_ |
