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Make an AudioEncoder subclass for Opus

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

Review URL: https://webrtc-codereview.appspot.com/23239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7552 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
kwiberg@webrtc.org
2014-10-29 07:28:36 +00:00
parent 2623695dfb
commit 663fdd02fd
6 changed files with 185 additions and 53 deletions
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69
webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
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@ -26,7 +26,7 @@
#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
#include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
#include "webrtc/system_wrappers/interface/data_log.h"
@ -140,17 +140,20 @@ class AudioDecoderTest : public ::testing::Test {
size_t input_len_samples,
uint8_t* output) {
size_t enc_len_bytes = 0;
scoped_ptr<int16_t[]> interleaved_input(
new int16_t[channels_ * input_len_samples]);
for (int i = 0; i < audio_encoder_->num_10ms_frames_per_packet(); ++i) {
EXPECT_EQ(0u, enc_len_bytes);
EXPECT_TRUE(audio_encoder_->Encode(0,
input,
audio_encoder_->sample_rate_hz() / 100,
data_length_ * 2,
output,
&enc_len_bytes,
&output_timestamp_));
// Duplicate the mono input signal to however many channels the test
// wants.
test::InputAudioFile::DuplicateInterleaved(
input, input_len_samples, channels_, interleaved_input.get());
EXPECT_TRUE(audio_encoder_->Encode(
0, interleaved_input.get(), audio_encoder_->sample_rate_hz() / 100,
data_length_ * 2, output, &enc_len_bytes, &output_timestamp_));
}
EXPECT_EQ(input_len_samples, enc_len_bytes);
return static_cast<int>(enc_len_bytes);
}
@ -636,56 +639,22 @@ class AudioDecoderOpusTest : public AudioDecoderTest {
frame_size_ = 480;
data_length_ = 10 * frame_size_;
decoder_ = new AudioDecoderOpus(kDecoderOpus);
assert(decoder_);
WebRtcOpus_EncoderCreate(&encoder_, 1);
AudioEncoderOpus::Config config;
config.frame_size_ms = static_cast<int>(frame_size_) / 48;
audio_encoder_.reset(new AudioEncoderOpus(config));
}
~AudioDecoderOpusTest() {
WebRtcOpus_EncoderFree(encoder_);
}
virtual void InitEncoder() {}
virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
uint8_t* output) OVERRIDE {
int enc_len_bytes = WebRtcOpus_Encode(encoder_, const_cast<int16_t*>(input),
static_cast<int16_t>(input_len_samples),
static_cast<int16_t>(data_length_), output);
EXPECT_GT(enc_len_bytes, 0);
return enc_len_bytes;
}
OpusEncInst* encoder_;
};
class AudioDecoderOpusStereoTest : public AudioDecoderOpusTest {
protected:
AudioDecoderOpusStereoTest() : AudioDecoderOpusTest() {
channels_ = 2;
WebRtcOpus_EncoderFree(encoder_);
delete decoder_;
decoder_ = new AudioDecoderOpus(kDecoderOpus_2ch);
assert(decoder_);
WebRtcOpus_EncoderCreate(&encoder_, 2);
}
virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
uint8_t* output) OVERRIDE {
// Create stereo by duplicating each sample in |input|.
const int input_stereo_samples = static_cast<int>(input_len_samples) * 2;
scoped_ptr<int16_t[]> input_stereo(new int16_t[input_stereo_samples]);
test::InputAudioFile::DuplicateInterleaved(
input, input_len_samples, 2, input_stereo.get());
// Note that the input length is given as samples per channel.
int enc_len_bytes =
WebRtcOpus_Encode(encoder_,
input_stereo.get(),
static_cast<int16_t>(input_len_samples),
static_cast<int16_t>(data_length_),
output);
EXPECT_GT(enc_len_bytes, 0);
return enc_len_bytes;
AudioEncoderOpus::Config config;
config.frame_size_ms = static_cast<int>(frame_size_) / 48;
config.num_channels = 2;
audio_encoder_.reset(new AudioEncoderOpus(config));
}
};