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AudioEncoder subclass for iSACfix

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This patch refactors AudioEncoderDecoderIsac so that it can share
almost all code with the very similar AudioEncoderDecoderIsacFix.

BUG=3926
R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7912 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
kwiberg@webrtc.org
2014-12-16 12:49:37 +00:00
parent 0198933b3d
commit 88bdec8c3a
12 changed files with 586 additions and 369 deletions
Download Patch File Download Diff File Expand all files Collapse all files

50
webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
View File

@ -23,7 +23,7 @@
#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
#endif
#ifdef WEBRTC_CODEC_ISACFX
#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h"
#endif
#ifdef WEBRTC_CODEC_ISAC
#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h"
@ -127,48 +127,6 @@ int AudioDecoderIlbc::Init() {
}
#endif
// iSAC fix
#ifdef WEBRTC_CODEC_ISACFX
AudioDecoderIsacFix::AudioDecoderIsacFix() {
WebRtcIsacfix_Create(&isac_state_);
}
AudioDecoderIsacFix::~AudioDecoderIsacFix() {
WebRtcIsacfix_Free(isac_state_);
}
int AudioDecoderIsacFix::Decode(const uint8_t* encoded, size_t encoded_len,
int16_t* decoded, SpeechType* speech_type) {
int16_t temp_type = 1; // Default is speech.
int16_t ret = WebRtcIsacfix_Decode(isac_state_,
encoded,
static_cast<int16_t>(encoded_len), decoded,
&temp_type);
*speech_type = ConvertSpeechType(temp_type);
return ret;
}
int AudioDecoderIsacFix::Init() {
return WebRtcIsacfix_DecoderInit(isac_state_);
}
int AudioDecoderIsacFix::IncomingPacket(const uint8_t* payload,
size_t payload_len,
uint16_t rtp_sequence_number,
uint32_t rtp_timestamp,
uint32_t arrival_timestamp) {
return WebRtcIsacfix_UpdateBwEstimate(
isac_state_,
payload,
static_cast<int32_t>(payload_len),
rtp_sequence_number, rtp_timestamp, arrival_timestamp);
}
int AudioDecoderIsacFix::ErrorCode() {
return WebRtcIsacfix_GetErrorCode(isac_state_);
}
#endif
// G.722
#ifdef WEBRTC_CODEC_G722
AudioDecoderG722::AudioDecoderG722() {
@ -485,8 +443,10 @@ AudioDecoder* CreateAudioDecoder(NetEqDecoder codec_type) {
return new AudioDecoderIlbc;
#endif
#if defined(WEBRTC_CODEC_ISACFX)
case kDecoderISAC:
return new AudioDecoderIsacFix;
case kDecoderISAC: {
AudioEncoderDecoderIsacFix::Config config;
return new AudioEncoderDecoderIsacFix(config);
}
#elif defined(WEBRTC_CODEC_ISAC)
case kDecoderISAC: {
AudioEncoderDecoderIsac::Config config;