| /* |
| * Copyright (c) 2015 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 "webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h" |
| |
| #include <string.h> |
| |
| #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h" |
| #include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h" |
| #include "webrtc/rtc_base/checks.h" |
| |
| namespace webrtc { |
| |
| AudioDecoderG722Impl::AudioDecoderG722Impl() { |
| WebRtcG722_CreateDecoder(&dec_state_); |
| WebRtcG722_DecoderInit(dec_state_); |
| } |
| |
| AudioDecoderG722Impl::~AudioDecoderG722Impl() { |
| WebRtcG722_FreeDecoder(dec_state_); |
| } |
| |
| bool AudioDecoderG722Impl::HasDecodePlc() const { |
| return false; |
| } |
| |
| int AudioDecoderG722Impl::DecodeInternal(const uint8_t* encoded, |
| size_t encoded_len, |
| int sample_rate_hz, |
| int16_t* decoded, |
| SpeechType* speech_type) { |
| RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz); |
| int16_t temp_type = 1; // Default is speech. |
| size_t ret = |
| WebRtcG722_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type); |
| *speech_type = ConvertSpeechType(temp_type); |
| return static_cast<int>(ret); |
| } |
| |
| void AudioDecoderG722Impl::Reset() { |
| WebRtcG722_DecoderInit(dec_state_); |
| } |
| |
| std::vector<AudioDecoder::ParseResult> AudioDecoderG722Impl::ParsePayload( |
| rtc::Buffer&& payload, |
| uint32_t timestamp) { |
| return LegacyEncodedAudioFrame::SplitBySamples(this, std::move(payload), |
| timestamp, 8, 16); |
| } |
| |
| int AudioDecoderG722Impl::PacketDuration(const uint8_t* encoded, |
| size_t encoded_len) const { |
| // 1/2 encoded byte per sample per channel. |
| return static_cast<int>(2 * encoded_len / Channels()); |
| } |
| |
| int AudioDecoderG722Impl::SampleRateHz() const { |
| return 16000; |
| } |
| |
| size_t AudioDecoderG722Impl::Channels() const { |
| return 1; |
| } |
| |
| AudioDecoderG722StereoImpl::AudioDecoderG722StereoImpl() { |
| WebRtcG722_CreateDecoder(&dec_state_left_); |
| WebRtcG722_CreateDecoder(&dec_state_right_); |
| WebRtcG722_DecoderInit(dec_state_left_); |
| WebRtcG722_DecoderInit(dec_state_right_); |
| } |
| |
| AudioDecoderG722StereoImpl::~AudioDecoderG722StereoImpl() { |
| WebRtcG722_FreeDecoder(dec_state_left_); |
| WebRtcG722_FreeDecoder(dec_state_right_); |
| } |
| |
| int AudioDecoderG722StereoImpl::DecodeInternal(const uint8_t* encoded, |
| size_t encoded_len, |
| int sample_rate_hz, |
| int16_t* decoded, |
| SpeechType* speech_type) { |
| RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz); |
| int16_t temp_type = 1; // Default is speech. |
| // De-interleave the bit-stream into two separate payloads. |
| uint8_t* encoded_deinterleaved = new uint8_t[encoded_len]; |
| SplitStereoPacket(encoded, encoded_len, encoded_deinterleaved); |
| // Decode left and right. |
| size_t decoded_len = WebRtcG722_Decode(dec_state_left_, encoded_deinterleaved, |
| encoded_len / 2, decoded, &temp_type); |
| size_t ret = WebRtcG722_Decode( |
| dec_state_right_, &encoded_deinterleaved[encoded_len / 2], |
| encoded_len / 2, &decoded[decoded_len], &temp_type); |
| if (ret == decoded_len) { |
| ret += decoded_len; // Return total number of samples. |
| // Interleave output. |
| for (size_t k = ret / 2; k < ret; k++) { |
| int16_t temp = decoded[k]; |
| memmove(&decoded[2 * k - ret + 2], &decoded[2 * k - ret + 1], |
| (ret - k - 1) * sizeof(int16_t)); |
| decoded[2 * k - ret + 1] = temp; |
| } |
| } |
| *speech_type = ConvertSpeechType(temp_type); |
| delete[] encoded_deinterleaved; |
| return static_cast<int>(ret); |
| } |
| |
| int AudioDecoderG722StereoImpl::SampleRateHz() const { |
| return 16000; |
| } |
| |
| size_t AudioDecoderG722StereoImpl::Channels() const { |
| return 2; |
| } |
| |
| void AudioDecoderG722StereoImpl::Reset() { |
| WebRtcG722_DecoderInit(dec_state_left_); |
| WebRtcG722_DecoderInit(dec_state_right_); |
| } |
| |
| std::vector<AudioDecoder::ParseResult> AudioDecoderG722StereoImpl::ParsePayload( |
| rtc::Buffer&& payload, |
| uint32_t timestamp) { |
| return LegacyEncodedAudioFrame::SplitBySamples(this, std::move(payload), |
| timestamp, 2 * 8, 16); |
| } |
| |
| // Split the stereo packet and place left and right channel after each other |
| // in the output array. |
| void AudioDecoderG722StereoImpl::SplitStereoPacket( |
| const uint8_t* encoded, |
| size_t encoded_len, |
| uint8_t* encoded_deinterleaved) { |
| // Regroup the 4 bits/sample so |l1 l2| |r1 r2| |l3 l4| |r3 r4| ..., |
| // where "lx" is 4 bits representing left sample number x, and "rx" right |
| // sample. Two samples fit in one byte, represented with |...|. |
| for (size_t i = 0; i + 1 < encoded_len; i += 2) { |
| uint8_t right_byte = ((encoded[i] & 0x0F) << 4) + (encoded[i + 1] & 0x0F); |
| encoded_deinterleaved[i] = (encoded[i] & 0xF0) + (encoded[i + 1] >> 4); |
| encoded_deinterleaved[i + 1] = right_byte; |
| } |
| |
| // Move one byte representing right channel each loop, and place it at the |
| // end of the bytestream vector. After looping the data is reordered to: |
| // |l1 l2| |l3 l4| ... |l(N-1) lN| |r1 r2| |r3 r4| ... |r(N-1) r(N)|, |
| // where N is the total number of samples. |
| for (size_t i = 0; i < encoded_len / 2; i++) { |
| uint8_t right_byte = encoded_deinterleaved[i + 1]; |
| memmove(&encoded_deinterleaved[i + 1], &encoded_deinterleaved[i + 2], |
| encoded_len - i - 2); |
| encoded_deinterleaved[encoded_len - 1] = right_byte; |
| } |
| } |
| |
| } // namespace webrtc |