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      1 /*
      2  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
      3  *
      4  *  Use of this source code is governed by a BSD-style license
      5  *  that can be found in the LICENSE file in the root of the source
      6  *  tree. An additional intellectual property rights grant can be found
      7  *  in the file PATENTS.  All contributing project authors may
      8  *  be found in the AUTHORS file in the root of the source tree.
      9  */
     10 
     11 #include "webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h"
     12 
     13 #include <string.h>
     14 
     15 #include "webrtc/base/checks.h"
     16 #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
     17 
     18 namespace webrtc {
     19 
     20 AudioDecoderG722::AudioDecoderG722() {
     21   WebRtcG722_CreateDecoder(&dec_state_);
     22   WebRtcG722_DecoderInit(dec_state_);
     23 }
     24 
     25 AudioDecoderG722::~AudioDecoderG722() {
     26   WebRtcG722_FreeDecoder(dec_state_);
     27 }
     28 
     29 bool AudioDecoderG722::HasDecodePlc() const {
     30   return false;
     31 }
     32 
     33 int AudioDecoderG722::DecodeInternal(const uint8_t* encoded,
     34                                      size_t encoded_len,
     35                                      int sample_rate_hz,
     36                                      int16_t* decoded,
     37                                      SpeechType* speech_type) {
     38   RTC_DCHECK_EQ(sample_rate_hz, 16000);
     39   int16_t temp_type = 1;  // Default is speech.
     40   size_t ret =
     41       WebRtcG722_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
     42   *speech_type = ConvertSpeechType(temp_type);
     43   return static_cast<int>(ret);
     44 }
     45 
     46 void AudioDecoderG722::Reset() {
     47   WebRtcG722_DecoderInit(dec_state_);
     48 }
     49 
     50 int AudioDecoderG722::PacketDuration(const uint8_t* encoded,
     51                                      size_t encoded_len) const {
     52   // 1/2 encoded byte per sample per channel.
     53   return static_cast<int>(2 * encoded_len / Channels());
     54 }
     55 
     56 size_t AudioDecoderG722::Channels() const {
     57   return 1;
     58 }
     59 
     60 AudioDecoderG722Stereo::AudioDecoderG722Stereo() {
     61   WebRtcG722_CreateDecoder(&dec_state_left_);
     62   WebRtcG722_CreateDecoder(&dec_state_right_);
     63   WebRtcG722_DecoderInit(dec_state_left_);
     64   WebRtcG722_DecoderInit(dec_state_right_);
     65 }
     66 
     67 AudioDecoderG722Stereo::~AudioDecoderG722Stereo() {
     68   WebRtcG722_FreeDecoder(dec_state_left_);
     69   WebRtcG722_FreeDecoder(dec_state_right_);
     70 }
     71 
     72 int AudioDecoderG722Stereo::DecodeInternal(const uint8_t* encoded,
     73                                            size_t encoded_len,
     74                                            int sample_rate_hz,
     75                                            int16_t* decoded,
     76                                            SpeechType* speech_type) {
     77   RTC_DCHECK_EQ(sample_rate_hz, 16000);
     78   int16_t temp_type = 1;  // Default is speech.
     79   // De-interleave the bit-stream into two separate payloads.
     80   uint8_t* encoded_deinterleaved = new uint8_t[encoded_len];
     81   SplitStereoPacket(encoded, encoded_len, encoded_deinterleaved);
     82   // Decode left and right.
     83   size_t decoded_len = WebRtcG722_Decode(dec_state_left_, encoded_deinterleaved,
     84                                          encoded_len / 2, decoded, &temp_type);
     85   size_t ret = WebRtcG722_Decode(
     86       dec_state_right_, &encoded_deinterleaved[encoded_len / 2],
     87       encoded_len / 2, &decoded[decoded_len], &temp_type);
     88   if (ret == decoded_len) {
     89     ret += decoded_len;  // Return total number of samples.
     90     // Interleave output.
     91     for (size_t k = ret / 2; k < ret; k++) {
     92       int16_t temp = decoded[k];
     93       memmove(&decoded[2 * k - ret + 2], &decoded[2 * k - ret + 1],
     94               (ret - k - 1) * sizeof(int16_t));
     95       decoded[2 * k - ret + 1] = temp;
     96     }
     97   }
     98   *speech_type = ConvertSpeechType(temp_type);
     99   delete[] encoded_deinterleaved;
    100   return static_cast<int>(ret);
    101 }
    102 
    103 size_t AudioDecoderG722Stereo::Channels() const {
    104   return 2;
    105 }
    106 
    107 void AudioDecoderG722Stereo::Reset() {
    108   WebRtcG722_DecoderInit(dec_state_left_);
    109   WebRtcG722_DecoderInit(dec_state_right_);
    110 }
    111 
    112 // Split the stereo packet and place left and right channel after each other
    113 // in the output array.
    114 void AudioDecoderG722Stereo::SplitStereoPacket(const uint8_t* encoded,
    115                                                size_t encoded_len,
    116                                                uint8_t* encoded_deinterleaved) {
    117   // Regroup the 4 bits/sample so |l1 l2| |r1 r2| |l3 l4| |r3 r4| ...,
    118   // where "lx" is 4 bits representing left sample number x, and "rx" right
    119   // sample. Two samples fit in one byte, represented with |...|.
    120   for (size_t i = 0; i + 1 < encoded_len; i += 2) {
    121     uint8_t right_byte = ((encoded[i] & 0x0F) << 4) + (encoded[i + 1] & 0x0F);
    122     encoded_deinterleaved[i] = (encoded[i] & 0xF0) + (encoded[i + 1] >> 4);
    123     encoded_deinterleaved[i + 1] = right_byte;
    124   }
    125 
    126   // Move one byte representing right channel each loop, and place it at the
    127   // end of the bytestream vector. After looping the data is reordered to:
    128   // |l1 l2| |l3 l4| ... |l(N-1) lN| |r1 r2| |r3 r4| ... |r(N-1) r(N)|,
    129   // where N is the total number of samples.
    130   for (size_t i = 0; i < encoded_len / 2; i++) {
    131     uint8_t right_byte = encoded_deinterleaved[i + 1];
    132     memmove(&encoded_deinterleaved[i + 1], &encoded_deinterleaved[i + 2],
    133             encoded_len - i - 2);
    134     encoded_deinterleaved[encoded_len - 1] = right_byte;
    135   }
    136 }
    137 
    138 }  // namespace webrtc
    139