Support for CELT in NetEq4.
BUG=1359 R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2291004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4884 4adac7df-926f-26a2-2b94-8c16560cd09d
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@ -74,6 +74,10 @@ bool AudioDecoder::CodecSupported(NetEqDecoder codec_type) {
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case kDecoderG722:
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case kDecoderG722_2ch:
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#endif
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#ifdef WEBRTC_CODEC_CELT
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case kDecoderCELT_32:
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case kDecoderCELT_32_2ch:
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#endif
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#ifdef WEBRTC_CODEC_OPUS
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case kDecoderOpus:
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case kDecoderOpus_2ch:
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@ -131,6 +135,10 @@ int AudioDecoder::CodecSampleRateHz(NetEqDecoder codec_type) {
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#ifdef WEBRTC_CODEC_PCM16
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case kDecoderPCM16Bswb32kHz:
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case kDecoderPCM16Bswb32kHz_2ch:
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#endif
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#ifdef WEBRTC_CODEC_CELT
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case kDecoderCELT_32:
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case kDecoderCELT_32_2ch:
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#endif
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case kDecoderCNGswb32kHz: {
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return 32000;
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@ -206,6 +214,11 @@ AudioDecoder* AudioDecoder::CreateAudioDecoder(NetEqDecoder codec_type) {
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case kDecoderG722_2ch:
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return new AudioDecoderG722Stereo;
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#endif
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#ifdef WEBRTC_CODEC_CELT
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case kDecoderCELT_32:
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case kDecoderCELT_32_2ch:
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return new AudioDecoderCelt(codec_type);
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#endif
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#ifdef WEBRTC_CODEC_OPUS
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case kDecoderOpus:
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case kDecoderOpus_2ch:
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@ -13,6 +13,9 @@
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#include <assert.h>
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#include <string.h> // memmove
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#ifdef WEBRTC_CODEC_CELT
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#include "webrtc/modules/audio_coding/codecs/celt/include/celt_interface.h"
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#endif
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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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@ -377,6 +380,55 @@ void AudioDecoderG722Stereo::SplitStereoPacket(const uint8_t* encoded,
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}
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#endif
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// CELT
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#ifdef WEBRTC_CODEC_CELT
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AudioDecoderCelt::AudioDecoderCelt(enum NetEqDecoder type)
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: AudioDecoder(type) {
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assert(type == kDecoderCELT_32 || type == kDecoderCELT_32_2ch);
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if (type == kDecoderCELT_32) {
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channels_ = 1;
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} else {
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channels_ = 2;
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}
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WebRtcCelt_CreateDec(reinterpret_cast<CELT_decinst_t**>(&state_),
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static_cast<int>(channels_));
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}
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AudioDecoderCelt::~AudioDecoderCelt() {
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WebRtcCelt_FreeDec(static_cast<CELT_decinst_t*>(state_));
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}
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int AudioDecoderCelt::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 to speech.
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int ret = WebRtcCelt_DecodeUniversal(static_cast<CELT_decinst_t*>(state_),
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encoded, static_cast<int>(encoded_len),
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decoded, &temp_type);
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*speech_type = ConvertSpeechType(temp_type);
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if (ret < 0) {
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return -1;
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}
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// Return the total number of samples.
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return ret * static_cast<int>(channels_);
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}
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int AudioDecoderCelt::Init() {
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return WebRtcCelt_DecoderInit(static_cast<CELT_decinst_t*>(state_));
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}
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bool AudioDecoderCelt::HasDecodePlc() const { return true; }
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int AudioDecoderCelt::DecodePlc(int num_frames, int16_t* decoded) {
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int ret = WebRtcCelt_DecodePlc(static_cast<CELT_decinst_t*>(state_),
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decoded, num_frames);
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if (ret < 0) {
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return -1;
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}
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// Return the total number of samples.
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return ret * static_cast<int>(channels_);
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}
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#endif
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// Opus
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#ifdef WEBRTC_CODEC_OPUS
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AudioDecoderOpus::AudioDecoderOpus(enum NetEqDecoder type)
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@ -212,6 +212,23 @@ class AudioDecoderG722Stereo : public AudioDecoderG722 {
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};
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#endif
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#ifdef WEBRTC_CODEC_CELT
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class AudioDecoderCelt : public AudioDecoder {
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public:
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explicit AudioDecoderCelt(enum NetEqDecoder type);
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virtual ~AudioDecoderCelt();
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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 bool HasDecodePlc() const;
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virtual int DecodePlc(int num_frames, int16_t* decoded);
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private:
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DISALLOW_COPY_AND_ASSIGN(AudioDecoderCelt);
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};
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#endif
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#ifdef WEBRTC_CODEC_OPUS
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class AudioDecoderOpus : public AudioDecoder {
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public:
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@ -17,6 +17,9 @@
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#include "gtest/gtest.h"
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#include "webrtc/common_audio/resampler/include/resampler.h"
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#ifdef WEBRTC_CODEC_CELT
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#include "webrtc/modules/audio_coding/codecs/celt/include/celt_interface.h"
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#endif
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#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
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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/ilbc/interface/ilbc.h"
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@ -143,11 +146,12 @@ class AudioDecoderTest : public ::testing::Test {
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// The absolute difference between the two channels in a stereo is compared vs
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// |tolerance|.
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virtual void CompareTwoChannels(size_t num_samples, int tolerance) const {
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assert(num_samples <= data_length_);
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for (unsigned int n = 0; n < num_samples; ++n)
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ASSERT_NEAR(decoded_[channels_ * n], decoded_[channels_ * n + 1],
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tolerance) << "Stereo samples differ.";
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virtual void CompareTwoChannels(size_t samples_per_channel,
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int tolerance) const {
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assert(samples_per_channel <= data_length_);
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for (unsigned int n = 0; n < samples_per_channel; ++n)
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ASSERT_NEAR(decoded_[channels_ * n], decoded_[channels_ * n + 1],
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tolerance) << "Stereo samples differ.";
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}
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// Calculates mean-squared error between input and output (the first channel).
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@ -199,12 +203,12 @@ class AudioDecoderTest : public ::testing::Test {
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EXPECT_EQ(0, decoder_->Init());
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size_t dec_len =
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decoder_->Decode(encoded_, enc_len, decoded_, &speech_type);
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EXPECT_EQ(frame_size_, dec_len);
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EXPECT_EQ(frame_size_ * channels_, dec_len);
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// Call DecodePlc and verify that we get one frame of data.
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// (Overwrite the output from the above Decode call, but that does not
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// matter.)
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dec_len = decoder_->DecodePlc(1, decoded_);
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EXPECT_EQ(frame_size_, dec_len);
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EXPECT_EQ(frame_size_ * channels_, dec_len);
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}
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std::string input_file_;
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@ -527,6 +531,77 @@ class AudioDecoderG722StereoTest : public AudioDecoderG722Test {
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}
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};
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#ifdef WEBRTC_CODEC_CELT
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class AudioDecoderCeltTest : public AudioDecoderTest {
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protected:
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static const int kEncodingRateBitsPerSecond = 64000;
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AudioDecoderCeltTest() : AudioDecoderTest(), encoder_(NULL) {
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frame_size_ = 640;
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data_length_ = 10 * frame_size_;
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decoder_ = AudioDecoder::CreateAudioDecoder(kDecoderCELT_32);
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assert(decoder_);
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WebRtcCelt_CreateEnc(&encoder_, static_cast<int>(channels_));
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}
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~AudioDecoderCeltTest() {
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WebRtcCelt_FreeEnc(encoder_);
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}
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virtual void InitEncoder() {
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assert(encoder_);
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ASSERT_EQ(0, WebRtcCelt_EncoderInit(
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encoder_, static_cast<int>(channels_), kEncodingRateBitsPerSecond));
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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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assert(encoder_);
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return WebRtcCelt_Encode(encoder_, input, output);
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}
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CELT_encinst_t* encoder_;
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};
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class AudioDecoderCeltStereoTest : public AudioDecoderTest {
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protected:
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static const int kEncodingRateBitsPerSecond = 64000;
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AudioDecoderCeltStereoTest() : AudioDecoderTest(), encoder_(NULL) {
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channels_ = 2;
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frame_size_ = 640;
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data_length_ = 10 * frame_size_;
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decoder_ = AudioDecoder::CreateAudioDecoder(kDecoderCELT_32_2ch);
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assert(decoder_);
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stereo_input_ = new int16_t[frame_size_ * channels_];
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WebRtcCelt_CreateEnc(&encoder_, static_cast<int>(channels_));
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}
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~AudioDecoderCeltStereoTest() {
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delete [] stereo_input_;
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WebRtcCelt_FreeEnc(encoder_);
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}
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virtual void InitEncoder() {
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assert(encoder_);
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ASSERT_EQ(0, WebRtcCelt_EncoderInit(
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encoder_, static_cast<int>(channels_), kEncodingRateBitsPerSecond));
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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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assert(encoder_);
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assert(stereo_input_);
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for (size_t n = 0; n < frame_size_; ++n) {
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stereo_input_[n * 2] = stereo_input_[n * 2 + 1] = input[n];
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}
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return WebRtcCelt_Encode(encoder_, stereo_input_, output);
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}
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int16_t* stereo_input_;
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CELT_encinst_t* encoder_;
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};
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#endif
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class AudioDecoderOpusTest : public AudioDecoderTest {
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protected:
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AudioDecoderOpusTest() : AudioDecoderTest() {
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@ -742,6 +817,38 @@ TEST_F(AudioDecoderOpusStereoTest, EncodeDecode) {
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EXPECT_FALSE(decoder_->HasDecodePlc());
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}
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#ifdef WEBRTC_CODEC_CELT
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// In the two following CELT tests, the low amplitude of the test signal allow
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// us to have such low error thresholds, i.e. |tolerance|, |mse|. Furthermore,
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// in general, stereo signals with identical channels do not result in identical
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// encoded channels.
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TEST_F(AudioDecoderCeltTest, EncodeDecode) {
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int tolerance = 20;
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double mse = 17.0;
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int delay = 80; // Delay from input to output in samples.
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EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderCELT_32));
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EncodeDecodeTest(1600, tolerance, mse, delay);
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ReInitTest();
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EXPECT_TRUE(decoder_->HasDecodePlc());
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DecodePlcTest();
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}
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TEST_F(AudioDecoderCeltStereoTest, EncodeDecode) {
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int tolerance = 20;
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// If both channels are identical, CELT not necessarily decodes identical
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// channels. However, for this input this is the case.
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int channel_diff_tolerance = 0;
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double mse = 20.0;
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// Delay from input to output in samples, accounting for stereo.
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int delay = 160;
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EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderCELT_32_2ch));
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EncodeDecodeTest(1600, tolerance, mse, delay, channel_diff_tolerance);
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ReInitTest();
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EXPECT_TRUE(decoder_->HasDecodePlc());
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DecodePlcTest();
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}
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#endif
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TEST(AudioDecoder, CodecSampleRateHz) {
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EXPECT_EQ(8000, AudioDecoder::CodecSampleRateHz(kDecoderPCMu));
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EXPECT_EQ(8000, AudioDecoder::CodecSampleRateHz(kDecoderPCMa));
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@ -772,8 +879,13 @@ TEST(AudioDecoder, CodecSampleRateHz) {
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EXPECT_EQ(-1, AudioDecoder::CodecSampleRateHz(kDecoderArbitrary));
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EXPECT_EQ(32000, AudioDecoder::CodecSampleRateHz(kDecoderOpus));
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EXPECT_EQ(32000, AudioDecoder::CodecSampleRateHz(kDecoderOpus_2ch));
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#ifdef WEBRTC_CODEC_CELT
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EXPECT_EQ(32000, AudioDecoder::CodecSampleRateHz(kDecoderCELT_32));
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EXPECT_EQ(32000, AudioDecoder::CodecSampleRateHz(kDecoderCELT_32_2ch));
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#else
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EXPECT_EQ(-1, AudioDecoder::CodecSampleRateHz(kDecoderCELT_32));
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EXPECT_EQ(-1, AudioDecoder::CodecSampleRateHz(kDecoderCELT_32_2ch));
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#endif
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}
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TEST(AudioDecoder, CodecSupported) {
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@ -805,8 +917,13 @@ TEST(AudioDecoder, CodecSupported) {
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EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderArbitrary));
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EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderOpus));
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EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderOpus_2ch));
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#ifdef WEBRTC_CODEC_CELT
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EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderCELT_32));
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EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderCELT_32_2ch));
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#else
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EXPECT_FALSE(AudioDecoder::CodecSupported(kDecoderCELT_32));
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EXPECT_FALSE(AudioDecoder::CodecSupported(kDecoderCELT_32_2ch));
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#endif
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}
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} // namespace webrtc
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