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/*
* 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 "modules/audio_coding/codecs/g722/audio_decoder_g722.h"
#include <string.h>
#include <utility>
#include "modules/audio_coding/codecs/g722/g722_interface.h"
#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
#include "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::PacketDurationRedundant(const uint8_t* encoded,
size_t encoded_len) const {
return PacketDuration(encoded, encoded_len);
}
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);
// Adjust the encoded length down to ensure the same number of samples in each
// channel.
const size_t encoded_len_adjusted = PacketDuration(encoded, encoded_len) *
Channels() /
2; // 1/2 byte per sample per channel
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_adjusted];
SplitStereoPacket(encoded, encoded_len_adjusted, encoded_deinterleaved);
// Decode left and right.
size_t decoded_len =
WebRtcG722_Decode(dec_state_left_, encoded_deinterleaved,
encoded_len_adjusted / 2, decoded, &temp_type);
size_t ret = WebRtcG722_Decode(
dec_state_right_, &encoded_deinterleaved[encoded_len_adjusted / 2],
encoded_len_adjusted / 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::PacketDuration(const uint8_t* encoded,
size_t encoded_len) const {
// 1/2 encoded byte per sample per channel. Make sure the length represents
// an equal number of bytes per channel. Otherwise, we cannot de-interleave
// the encoded data later.
return static_cast<int>(2 * (encoded_len / Channels()));
}
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