AudioEncoder subclass for iSACfix
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
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@ -23,7 +23,7 @@
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#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
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#endif
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#ifdef WEBRTC_CODEC_ISACFX
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#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
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#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h"
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#endif
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#ifdef WEBRTC_CODEC_ISAC
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#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h"
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@ -127,48 +127,6 @@ int AudioDecoderIlbc::Init() {
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}
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#endif
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// iSAC fix
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#ifdef WEBRTC_CODEC_ISACFX
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AudioDecoderIsacFix::AudioDecoderIsacFix() {
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WebRtcIsacfix_Create(&isac_state_);
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}
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AudioDecoderIsacFix::~AudioDecoderIsacFix() {
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WebRtcIsacfix_Free(isac_state_);
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}
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int AudioDecoderIsacFix::Decode(const uint8_t* encoded, size_t encoded_len,
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int16_t* decoded, SpeechType* speech_type) {
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int16_t temp_type = 1; // Default is speech.
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int16_t ret = WebRtcIsacfix_Decode(isac_state_,
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encoded,
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static_cast<int16_t>(encoded_len), decoded,
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&temp_type);
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*speech_type = ConvertSpeechType(temp_type);
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return ret;
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}
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int AudioDecoderIsacFix::Init() {
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return WebRtcIsacfix_DecoderInit(isac_state_);
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}
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int AudioDecoderIsacFix::IncomingPacket(const uint8_t* payload,
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size_t payload_len,
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uint16_t rtp_sequence_number,
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uint32_t rtp_timestamp,
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uint32_t arrival_timestamp) {
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return WebRtcIsacfix_UpdateBwEstimate(
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isac_state_,
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payload,
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static_cast<int32_t>(payload_len),
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rtp_sequence_number, rtp_timestamp, arrival_timestamp);
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}
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int AudioDecoderIsacFix::ErrorCode() {
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return WebRtcIsacfix_GetErrorCode(isac_state_);
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}
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#endif
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// G.722
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#ifdef WEBRTC_CODEC_G722
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AudioDecoderG722::AudioDecoderG722() {
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@ -485,8 +443,10 @@ AudioDecoder* CreateAudioDecoder(NetEqDecoder codec_type) {
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return new AudioDecoderIlbc;
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#endif
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#if defined(WEBRTC_CODEC_ISACFX)
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case kDecoderISAC:
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return new AudioDecoderIsacFix;
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case kDecoderISAC: {
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AudioEncoderDecoderIsacFix::Config config;
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return new AudioEncoderDecoderIsacFix(config);
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}
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#elif defined(WEBRTC_CODEC_ISAC)
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case kDecoderISAC: {
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AudioEncoderDecoderIsac::Config config;
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@ -27,12 +27,6 @@
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#ifdef WEBRTC_CODEC_ILBC
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#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
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#endif
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#ifdef WEBRTC_CODEC_ISACFX
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#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
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#endif
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#ifdef WEBRTC_CODEC_ISAC
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#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
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#endif
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#ifdef WEBRTC_CODEC_OPUS
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#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
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#endif
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@ -130,27 +124,6 @@ class AudioDecoderIlbc : public AudioDecoder {
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};
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#endif
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#ifdef WEBRTC_CODEC_ISACFX
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class AudioDecoderIsacFix : public AudioDecoder {
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public:
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AudioDecoderIsacFix();
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virtual ~AudioDecoderIsacFix();
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virtual int Decode(const uint8_t* encoded, size_t encoded_len,
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int16_t* decoded, SpeechType* speech_type);
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virtual int Init();
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virtual int IncomingPacket(const uint8_t* payload,
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size_t payload_len,
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uint16_t rtp_sequence_number,
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uint32_t rtp_timestamp,
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uint32_t arrival_timestamp);
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virtual int ErrorCode();
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private:
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ISACFIX_MainStruct* isac_state_;
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DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacFix);
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};
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#endif
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#ifdef WEBRTC_CODEC_G722
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class AudioDecoderG722 : public AudioDecoder {
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public:
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@ -20,7 +20,7 @@
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#include "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h"
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#include "webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h"
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#include "webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h"
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#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
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#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h"
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#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h"
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#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"
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#include "webrtc/modules/audio_coding/codecs/pcm16b/include/audio_encoder_pcm16b.h"
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@ -388,38 +388,20 @@ class AudioDecoderIsacFixTest : public AudioDecoderTest {
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protected:
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AudioDecoderIsacFixTest() : AudioDecoderTest() {
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codec_input_rate_hz_ = 16000;
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input_size_ = 160;
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frame_size_ = 480;
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data_length_ = 10 * frame_size_;
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decoder_ = new AudioDecoderIsacFix;
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assert(decoder_);
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WebRtcIsacfix_Create(&encoder_);
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}
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AudioEncoderDecoderIsacFix::Config config;
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config.payload_type = payload_type_;
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config.sample_rate_hz = codec_input_rate_hz_;
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config.frame_size_ms =
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1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
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~AudioDecoderIsacFixTest() {
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WebRtcIsacfix_Free(encoder_);
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// We need to create separate AudioEncoderDecoderIsacFix objects for
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// encoding and decoding, because the test class destructor destroys them
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// both.
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audio_encoder_.reset(new AudioEncoderDecoderIsacFix(config));
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decoder_ = new AudioEncoderDecoderIsacFix(config);
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}
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virtual void InitEncoder() {
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ASSERT_EQ(0, WebRtcIsacfix_EncoderInit(encoder_, 1)); // Fixed mode.
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ASSERT_EQ(0,
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WebRtcIsacfix_Control(encoder_, 32000, 30)); // 32 kbps, 30 ms.
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}
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virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
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uint8_t* output) {
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// Insert 3 * 10 ms. Expect non-zero output on third call.
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EXPECT_EQ(0, WebRtcIsacfix_Encode(encoder_, input, output));
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input += input_size_;
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EXPECT_EQ(0, WebRtcIsacfix_Encode(encoder_, input, output));
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input += input_size_;
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int enc_len_bytes = WebRtcIsacfix_Encode(encoder_, input, output);
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EXPECT_GT(enc_len_bytes, 0);
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return enc_len_bytes;
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}
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ISACFIX_MainStruct* encoder_;
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int input_size_;
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};
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class AudioDecoderG722Test : public AudioDecoderTest {
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