Move AudioDecoderG722 next to AudioEncoderG722
All AudioDecoder subclasses have historically lived in NetEq, but they fit better with the codec they wrap. BUG=webrtc:4557 Review URL: https://codereview.webrtc.org/1346993002 Cr-Commit-Position: refs/heads/master@{#9966}
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kwiberg
@ -11,13 +11,12 @@
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#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
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#include <assert.h>
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#include <string.h> // memmove
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#include "webrtc/base/checks.h"
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#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
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#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
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#ifdef WEBRTC_CODEC_G722
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#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
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#include "webrtc/modules/audio_coding/codecs/g722/include/audio_decoder_g722.h"
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#endif
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#ifdef WEBRTC_CODEC_ILBC
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#include "webrtc/modules/audio_coding/codecs/ilbc/interface/audio_decoder_ilbc.h"
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@ -95,128 +94,6 @@ size_t AudioDecoderPcmAMultiCh::Channels() const {
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return channels_;
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}
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// G.722
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#ifdef WEBRTC_CODEC_G722
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AudioDecoderG722::AudioDecoderG722() {
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WebRtcG722_CreateDecoder(&dec_state_);
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WebRtcG722_DecoderInit(dec_state_);
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}
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AudioDecoderG722::~AudioDecoderG722() {
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WebRtcG722_FreeDecoder(dec_state_);
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}
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bool AudioDecoderG722::HasDecodePlc() const {
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return false;
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}
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int AudioDecoderG722::DecodeInternal(const uint8_t* encoded,
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size_t encoded_len,
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int sample_rate_hz,
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int16_t* decoded,
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SpeechType* speech_type) {
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RTC_DCHECK_EQ(sample_rate_hz, 16000);
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int16_t temp_type = 1; // Default is speech.
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size_t ret =
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WebRtcG722_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
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*speech_type = ConvertSpeechType(temp_type);
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return static_cast<int>(ret);
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}
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void AudioDecoderG722::Reset() {
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WebRtcG722_DecoderInit(dec_state_);
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}
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int AudioDecoderG722::PacketDuration(const uint8_t* encoded,
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size_t encoded_len) const {
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// 1/2 encoded byte per sample per channel.
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return static_cast<int>(2 * encoded_len / Channels());
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}
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size_t AudioDecoderG722::Channels() const {
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return 1;
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}
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AudioDecoderG722Stereo::AudioDecoderG722Stereo() {
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WebRtcG722_CreateDecoder(&dec_state_left_);
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WebRtcG722_CreateDecoder(&dec_state_right_);
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WebRtcG722_DecoderInit(dec_state_left_);
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WebRtcG722_DecoderInit(dec_state_right_);
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}
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AudioDecoderG722Stereo::~AudioDecoderG722Stereo() {
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WebRtcG722_FreeDecoder(dec_state_left_);
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WebRtcG722_FreeDecoder(dec_state_right_);
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}
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int AudioDecoderG722Stereo::DecodeInternal(const uint8_t* encoded,
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size_t encoded_len,
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int sample_rate_hz,
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int16_t* decoded,
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SpeechType* speech_type) {
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RTC_DCHECK_EQ(sample_rate_hz, 16000);
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int16_t temp_type = 1; // Default is speech.
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// De-interleave the bit-stream into two separate payloads.
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uint8_t* encoded_deinterleaved = new uint8_t[encoded_len];
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SplitStereoPacket(encoded, encoded_len, encoded_deinterleaved);
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// Decode left and right.
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size_t decoded_len = WebRtcG722_Decode(dec_state_left_, encoded_deinterleaved,
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encoded_len / 2, decoded, &temp_type);
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size_t ret = WebRtcG722_Decode(
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dec_state_right_, &encoded_deinterleaved[encoded_len / 2],
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encoded_len / 2, &decoded[decoded_len], &temp_type);
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if (ret == decoded_len) {
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ret += decoded_len; // Return total number of samples.
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// Interleave output.
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for (size_t k = ret / 2; k < ret; k++) {
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int16_t temp = decoded[k];
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memmove(&decoded[2 * k - ret + 2], &decoded[2 * k - ret + 1],
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(ret - k - 1) * sizeof(int16_t));
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decoded[2 * k - ret + 1] = temp;
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}
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}
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*speech_type = ConvertSpeechType(temp_type);
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delete [] encoded_deinterleaved;
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return static_cast<int>(ret);
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}
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size_t AudioDecoderG722Stereo::Channels() const {
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return 2;
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}
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void AudioDecoderG722Stereo::Reset() {
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WebRtcG722_DecoderInit(dec_state_left_);
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WebRtcG722_DecoderInit(dec_state_right_);
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}
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// Split the stereo packet and place left and right channel after each other
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// in the output array.
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void AudioDecoderG722Stereo::SplitStereoPacket(const uint8_t* encoded,
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size_t encoded_len,
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uint8_t* encoded_deinterleaved) {
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assert(encoded);
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// Regroup the 4 bits/sample so |l1 l2| |r1 r2| |l3 l4| |r3 r4| ...,
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// where "lx" is 4 bits representing left sample number x, and "rx" right
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// sample. Two samples fit in one byte, represented with |...|.
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for (size_t i = 0; i + 1 < encoded_len; i += 2) {
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uint8_t right_byte = ((encoded[i] & 0x0F) << 4) + (encoded[i + 1] & 0x0F);
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encoded_deinterleaved[i] = (encoded[i] & 0xF0) + (encoded[i + 1] >> 4);
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encoded_deinterleaved[i + 1] = right_byte;
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}
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// Move one byte representing right channel each loop, and place it at the
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// end of the bytestream vector. After looping the data is reordered to:
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// |l1 l2| |l3 l4| ... |l(N-1) lN| |r1 r2| |r3 r4| ... |r(N-1) r(N)|,
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// where N is the total number of samples.
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for (size_t i = 0; i < encoded_len / 2; i++) {
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uint8_t right_byte = encoded_deinterleaved[i + 1];
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memmove(&encoded_deinterleaved[i + 1], &encoded_deinterleaved[i + 2],
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encoded_len - i - 2);
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encoded_deinterleaved[encoded_len - 1] = right_byte;
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}
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}
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#endif
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AudioDecoderCng::AudioDecoderCng() {
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RTC_CHECK_EQ(0, WebRtcCng_CreateDec(&dec_state_));
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WebRtcCng_InitDec(dec_state_);
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