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Make an AudioEncoder subclass for iSAC

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BUG=3926

Previously committed: https://code.google.com/p/webrtc/source/detail?r=7675
and reverted: https://code.google.com/p/webrtc/source/detail?r=7676

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

Review URL: https://webrtc-codereview.appspot.com/25359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7871 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
kwiberg@webrtc.org
2014-12-11 10:08:19 +00:00
parent abe3f1879c
commit b3ad8cf6ca
7 changed files with 357 additions and 149 deletions
Download Patch File Download Diff File Expand all files Collapse all files

77
webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
View File

@ -26,7 +26,7 @@
#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
#endif
#ifdef WEBRTC_CODEC_ISAC
#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h"
#endif
#ifdef WEBRTC_CODEC_OPUS
#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
@ -127,67 +127,6 @@ int AudioDecoderIlbc::Init() {
}
#endif
// iSAC float
#ifdef WEBRTC_CODEC_ISAC
AudioDecoderIsac::AudioDecoderIsac(int decode_sample_rate_hz) {
DCHECK(decode_sample_rate_hz == 16000 || decode_sample_rate_hz == 32000);
WebRtcIsac_Create(&isac_state_);
WebRtcIsac_SetDecSampRate(isac_state_, decode_sample_rate_hz);
}
AudioDecoderIsac::~AudioDecoderIsac() {
WebRtcIsac_Free(isac_state_);
}
int AudioDecoderIsac::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 = WebRtcIsac_Decode(isac_state_,
encoded,
static_cast<int16_t>(encoded_len), decoded,
&temp_type);
*speech_type = ConvertSpeechType(temp_type);
return ret;
}
int AudioDecoderIsac::DecodeRedundant(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 = WebRtcIsac_DecodeRcu(isac_state_,
encoded,
static_cast<int16_t>(encoded_len), decoded,
&temp_type);
*speech_type = ConvertSpeechType(temp_type);
return ret;
}
int AudioDecoderIsac::DecodePlc(int num_frames, int16_t* decoded) {
return WebRtcIsac_DecodePlc(isac_state_, decoded, num_frames);
}
int AudioDecoderIsac::Init() {
return WebRtcIsac_DecoderInit(isac_state_);
}
int AudioDecoderIsac::IncomingPacket(const uint8_t* payload,
size_t payload_len,
uint16_t rtp_sequence_number,
uint32_t rtp_timestamp,
uint32_t arrival_timestamp) {
return WebRtcIsac_UpdateBwEstimate(isac_state_,
payload,
static_cast<int32_t>(payload_len),
rtp_sequence_number,
rtp_timestamp,
arrival_timestamp);
}
int AudioDecoderIsac::ErrorCode() {
return WebRtcIsac_GetErrorCode(isac_state_);
}
#endif
// iSAC fix
#ifdef WEBRTC_CODEC_ISACFX
AudioDecoderIsacFix::AudioDecoderIsacFix() {
@ -549,11 +488,17 @@ AudioDecoder* CreateAudioDecoder(NetEqDecoder codec_type) {
case kDecoderISAC:
return new AudioDecoderIsacFix;
#elif defined(WEBRTC_CODEC_ISAC)
case kDecoderISAC:
return new AudioDecoderIsac(16000);
case kDecoderISAC: {
AudioEncoderDecoderIsac::Config config;
config.sample_rate_hz = 16000;
return new AudioEncoderDecoderIsac(config);
}
case kDecoderISACswb:
case kDecoderISACfb:
return new AudioDecoderIsac(32000);
case kDecoderISACfb: {
AudioEncoderDecoderIsac::Config config;
config.sample_rate_hz = 32000;
return new AudioEncoderDecoderIsac(config);
}
#endif
#ifdef WEBRTC_CODEC_PCM16
case kDecoderPCM16B: