Remove CELT support from audio_coding.

R=henrik.lundin@webrtc.org, juberti@webrtc.org
TBR=kjellander@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/33579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7864 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org
2014-12-10 11:49:13 +00:00
parent 8084f9500f
commit d8ca723de7
14 changed files with 3 additions and 623 deletions

View File

@ -14,9 +14,6 @@
#include <string.h> // memmove
#include "webrtc/base/checks.h"
#ifdef WEBRTC_CODEC_CELT
#include "webrtc/modules/audio_coding/codecs/celt/include/celt_interface.h"
#endif
#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
#ifdef WEBRTC_CODEC_G722
@ -345,50 +342,6 @@ void AudioDecoderG722Stereo::SplitStereoPacket(const uint8_t* encoded,
}
#endif
// CELT
#ifdef WEBRTC_CODEC_CELT
AudioDecoderCelt::AudioDecoderCelt(int num_channels) {
DCHECK(num_channels == 1 || num_channels == 2);
channels_ = num_channels;
WebRtcCelt_CreateDec(reinterpret_cast<CELT_decinst_t**>(&state_),
static_cast<int>(channels_));
}
AudioDecoderCelt::~AudioDecoderCelt() {
WebRtcCelt_FreeDec(static_cast<CELT_decinst_t*>(state_));
}
int AudioDecoderCelt::Decode(const uint8_t* encoded, size_t encoded_len,
int16_t* decoded, SpeechType* speech_type) {
int16_t temp_type = 1; // Default to speech.
int ret = WebRtcCelt_DecodeUniversal(static_cast<CELT_decinst_t*>(state_),
encoded, static_cast<int>(encoded_len),
decoded, &temp_type);
*speech_type = ConvertSpeechType(temp_type);
if (ret < 0) {
return -1;
}
// Return the total number of samples.
return ret * static_cast<int>(channels_);
}
int AudioDecoderCelt::Init() {
return WebRtcCelt_DecoderInit(static_cast<CELT_decinst_t*>(state_));
}
bool AudioDecoderCelt::HasDecodePlc() const { return true; }
int AudioDecoderCelt::DecodePlc(int num_frames, int16_t* decoded) {
int ret = WebRtcCelt_DecodePlc(static_cast<CELT_decinst_t*>(state_),
decoded, num_frames);
if (ret < 0) {
return -1;
}
// Return the total number of samples.
return ret * static_cast<int>(channels_);
}
#endif
// Opus
#ifdef WEBRTC_CODEC_OPUS
AudioDecoderOpus::AudioDecoderOpus(int num_channels) {
@ -492,10 +445,6 @@ bool CodecSupported(NetEqDecoder codec_type) {
case kDecoderG722:
case kDecoderG722_2ch:
#endif
#ifdef WEBRTC_CODEC_CELT
case kDecoderCELT_32:
case kDecoderCELT_32_2ch:
#endif
#ifdef WEBRTC_CODEC_OPUS
case kDecoderOpus:
case kDecoderOpus_2ch:
@ -553,10 +502,6 @@ int CodecSampleRateHz(NetEqDecoder codec_type) {
#ifdef WEBRTC_CODEC_PCM16
case kDecoderPCM16Bswb32kHz:
case kDecoderPCM16Bswb32kHz_2ch:
#endif
#ifdef WEBRTC_CODEC_CELT
case kDecoderCELT_32:
case kDecoderCELT_32_2ch:
#endif
case kDecoderCNGswb32kHz: {
return 32000;
@ -630,12 +575,6 @@ AudioDecoder* CreateAudioDecoder(NetEqDecoder codec_type) {
case kDecoderG722_2ch:
return new AudioDecoderG722Stereo;
#endif
#ifdef WEBRTC_CODEC_CELT
case kDecoderCELT_32:
return new AudioDecoderCelt(1);
case kDecoderCELT_32_2ch:
return new AudioDecoderCelt(2);
#endif
#ifdef WEBRTC_CODEC_OPUS
case kDecoderOpus:
return new AudioDecoderOpus(1);