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Moving src/webrtc into src/.

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In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
This commit is contained in:
Mirko Bonadei
2017-09-15 06:15:48 +02:00
committed by Commit Bot
parent 6674846b4a
commit bb547203bf
4576 changed files with 1092 additions and 1196 deletions
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390
modules/audio_coding/test/opus_test.cc Normal file
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/*
* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_coding/test/opus_test.h"
#include <assert.h>
#include <string>
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "webrtc/modules/audio_coding/test/TestStereo.h"
#include "webrtc/modules/audio_coding/test/utility.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/typedefs.h"
namespace webrtc {
OpusTest::OpusTest()
: acm_receiver_(AudioCodingModule::Create(0)),
channel_a2b_(NULL),
counter_(0),
payload_type_(255),
rtp_timestamp_(0) {}
OpusTest::~OpusTest() {
if (channel_a2b_ != NULL) {
delete channel_a2b_;
channel_a2b_ = NULL;
}
if (opus_mono_encoder_ != NULL) {
WebRtcOpus_EncoderFree(opus_mono_encoder_);
opus_mono_encoder_ = NULL;
}
if (opus_stereo_encoder_ != NULL) {
WebRtcOpus_EncoderFree(opus_stereo_encoder_);
opus_stereo_encoder_ = NULL;
}
if (opus_mono_decoder_ != NULL) {
WebRtcOpus_DecoderFree(opus_mono_decoder_);
opus_mono_decoder_ = NULL;
}
if (opus_stereo_decoder_ != NULL) {
WebRtcOpus_DecoderFree(opus_stereo_decoder_);
opus_stereo_decoder_ = NULL;
}
}
void OpusTest::Perform() {
#ifndef WEBRTC_CODEC_OPUS
// Opus isn't defined, exit.
return;
#else
uint16_t frequency_hz;
size_t audio_channels;
int16_t test_cntr = 0;
// Open both mono and stereo test files in 32 kHz.
const std::string file_name_stereo =
webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm");
const std::string file_name_mono =
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
frequency_hz = 32000;
in_file_stereo_.Open(file_name_stereo, frequency_hz, "rb");
in_file_stereo_.ReadStereo(true);
in_file_mono_.Open(file_name_mono, frequency_hz, "rb");
in_file_mono_.ReadStereo(false);
// Create Opus encoders for mono and stereo.
ASSERT_GT(WebRtcOpus_EncoderCreate(&opus_mono_encoder_, 1, 0), -1);
ASSERT_GT(WebRtcOpus_EncoderCreate(&opus_stereo_encoder_, 2, 1), -1);
// Create Opus decoders for mono and stereo for stand-alone testing of Opus.
ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_mono_decoder_, 1), -1);
ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_stereo_decoder_, 2), -1);
WebRtcOpus_DecoderInit(opus_mono_decoder_);
WebRtcOpus_DecoderInit(opus_stereo_decoder_);
ASSERT_TRUE(acm_receiver_.get() != NULL);
EXPECT_EQ(0, acm_receiver_->InitializeReceiver());
// Register Opus stereo as receiving codec.
CodecInst opus_codec_param;
int codec_id = acm_receiver_->Codec("opus", 48000, 2);
EXPECT_EQ(0, acm_receiver_->Codec(codec_id, &opus_codec_param));
payload_type_ = opus_codec_param.pltype;
EXPECT_EQ(true,
acm_receiver_->RegisterReceiveCodec(
opus_codec_param.pltype, CodecInstToSdp(opus_codec_param)));
// Create and connect the channel.
channel_a2b_ = new TestPackStereo;
channel_a2b_->RegisterReceiverACM(acm_receiver_.get());
//
// Test Stereo.
//
channel_a2b_->set_codec_mode(kStereo);
audio_channels = 2;
test_cntr++;
OpenOutFile(test_cntr);
// Run Opus with 2.5 ms frame size.
Run(channel_a2b_, audio_channels, 64000, 120);
// Run Opus with 5 ms frame size.
Run(channel_a2b_, audio_channels, 64000, 240);
// Run Opus with 10 ms frame size.
Run(channel_a2b_, audio_channels, 64000, 480);
// Run Opus with 20 ms frame size.
Run(channel_a2b_, audio_channels, 64000, 960);
// Run Opus with 40 ms frame size.
Run(channel_a2b_, audio_channels, 64000, 1920);
// Run Opus with 60 ms frame size.
Run(channel_a2b_, audio_channels, 64000, 2880);
out_file_.Close();
out_file_standalone_.Close();
//
// Test Opus stereo with packet-losses.
//
test_cntr++;
OpenOutFile(test_cntr);
// Run Opus with 20 ms frame size, 1% packet loss.
Run(channel_a2b_, audio_channels, 64000, 960, 1);
// Run Opus with 20 ms frame size, 5% packet loss.
Run(channel_a2b_, audio_channels, 64000, 960, 5);
// Run Opus with 20 ms frame size, 10% packet loss.
Run(channel_a2b_, audio_channels, 64000, 960, 10);
out_file_.Close();
out_file_standalone_.Close();
//
// Test Mono.
//
channel_a2b_->set_codec_mode(kMono);
audio_channels = 1;
test_cntr++;
OpenOutFile(test_cntr);
// Register Opus mono as receiving codec.
opus_codec_param.channels = 1;
EXPECT_EQ(true,
acm_receiver_->RegisterReceiveCodec(
opus_codec_param.pltype, CodecInstToSdp(opus_codec_param)));
// Run Opus with 2.5 ms frame size.
Run(channel_a2b_, audio_channels, 32000, 120);
// Run Opus with 5 ms frame size.
Run(channel_a2b_, audio_channels, 32000, 240);
// Run Opus with 10 ms frame size.
Run(channel_a2b_, audio_channels, 32000, 480);
// Run Opus with 20 ms frame size.
Run(channel_a2b_, audio_channels, 32000, 960);
// Run Opus with 40 ms frame size.
Run(channel_a2b_, audio_channels, 32000, 1920);
// Run Opus with 60 ms frame size.
Run(channel_a2b_, audio_channels, 32000, 2880);
out_file_.Close();
out_file_standalone_.Close();
//
// Test Opus mono with packet-losses.
//
test_cntr++;
OpenOutFile(test_cntr);
// Run Opus with 20 ms frame size, 1% packet loss.
Run(channel_a2b_, audio_channels, 64000, 960, 1);
// Run Opus with 20 ms frame size, 5% packet loss.
Run(channel_a2b_, audio_channels, 64000, 960, 5);
// Run Opus with 20 ms frame size, 10% packet loss.
Run(channel_a2b_, audio_channels, 64000, 960, 10);
// Close the files.
in_file_stereo_.Close();
in_file_mono_.Close();
out_file_.Close();
out_file_standalone_.Close();
#endif
}
void OpusTest::Run(TestPackStereo* channel, size_t channels, int bitrate,
size_t frame_length, int percent_loss) {
AudioFrame audio_frame;
int32_t out_freq_hz_b = out_file_.SamplingFrequency();
const size_t kBufferSizeSamples = 480 * 12 * 2; // 120 ms stereo audio.
int16_t audio[kBufferSizeSamples];
int16_t out_audio[kBufferSizeSamples];
int16_t audio_type;
size_t written_samples = 0;
size_t read_samples = 0;
size_t decoded_samples = 0;
bool first_packet = true;
uint32_t start_time_stamp = 0;
channel->reset_payload_size();
counter_ = 0;
// Set encoder rate.
EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_mono_encoder_, bitrate));
EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_stereo_encoder_, bitrate));
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
// If we are on Android, iOS and/or ARM, use a lower complexity setting as
// default.
const int kOpusComplexity5 = 5;
EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_mono_encoder_, kOpusComplexity5));
EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_stereo_encoder_,
kOpusComplexity5));
#endif
// Fast-forward 1 second (100 blocks) since the files start with silence.
in_file_stereo_.FastForward(100);
in_file_mono_.FastForward(100);
// Limit the runtime to 1000 blocks of 10 ms each.
for (size_t audio_length = 0; audio_length < 1000; audio_length += 10) {
bool lost_packet = false;
// Get 10 msec of audio.
if (channels == 1) {
if (in_file_mono_.EndOfFile()) {
break;
}
in_file_mono_.Read10MsData(audio_frame);
} else {
if (in_file_stereo_.EndOfFile()) {
break;
}
in_file_stereo_.Read10MsData(audio_frame);
}
// If input audio is sampled at 32 kHz, resampling to 48 kHz is required.
EXPECT_EQ(480,
resampler_.Resample10Msec(audio_frame.data(),
audio_frame.sample_rate_hz_,
48000,
channels,
kBufferSizeSamples - written_samples,
&audio[written_samples]));
written_samples += 480 * channels;
// Sometimes we need to loop over the audio vector to produce the right
// number of packets.
size_t loop_encode = (written_samples - read_samples) /
(channels * frame_length);
if (loop_encode > 0) {
const size_t kMaxBytes = 1000; // Maximum number of bytes for one packet.
size_t bitstream_len_byte;
uint8_t bitstream[kMaxBytes];
for (size_t i = 0; i < loop_encode; i++) {
int bitstream_len_byte_int = WebRtcOpus_Encode(
(channels == 1) ? opus_mono_encoder_ : opus_stereo_encoder_,
&audio[read_samples], frame_length, kMaxBytes, bitstream);
ASSERT_GE(bitstream_len_byte_int, 0);
bitstream_len_byte = static_cast<size_t>(bitstream_len_byte_int);
// Simulate packet loss by setting |packet_loss_| to "true" in
// |percent_loss| percent of the loops.
// TODO(tlegrand): Move handling of loss simulation to TestPackStereo.
if (percent_loss > 0) {
if (counter_ == floor((100 / percent_loss) + 0.5)) {
counter_ = 0;
lost_packet = true;
channel->set_lost_packet(true);
} else {
lost_packet = false;
channel->set_lost_packet(false);
}
counter_++;
}
// Run stand-alone Opus decoder, or decode PLC.
if (channels == 1) {
if (!lost_packet) {
decoded_samples += WebRtcOpus_Decode(
opus_mono_decoder_, bitstream, bitstream_len_byte,
&out_audio[decoded_samples * channels], &audio_type);
} else {
decoded_samples += WebRtcOpus_DecodePlc(
opus_mono_decoder_, &out_audio[decoded_samples * channels], 1);
}
} else {
if (!lost_packet) {
decoded_samples += WebRtcOpus_Decode(
opus_stereo_decoder_, bitstream, bitstream_len_byte,
&out_audio[decoded_samples * channels], &audio_type);
} else {
decoded_samples += WebRtcOpus_DecodePlc(
opus_stereo_decoder_, &out_audio[decoded_samples * channels],
1);
}
}
// Send data to the channel. "channel" will handle the loss simulation.
channel->SendData(kAudioFrameSpeech, payload_type_, rtp_timestamp_,
bitstream, bitstream_len_byte, NULL);
if (first_packet) {
first_packet = false;
start_time_stamp = rtp_timestamp_;
}
rtp_timestamp_ += static_cast<uint32_t>(frame_length);
read_samples += frame_length * channels;
}
if (read_samples == written_samples) {
read_samples = 0;
written_samples = 0;
}
}
// Run received side of ACM.
bool muted;
ASSERT_EQ(
0, acm_receiver_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted));
ASSERT_FALSE(muted);
// Write output speech to file.
out_file_.Write10MsData(
audio_frame.data(),
audio_frame.samples_per_channel_ * audio_frame.num_channels_);
// Write stand-alone speech to file.
out_file_standalone_.Write10MsData(out_audio, decoded_samples * channels);
if (audio_frame.timestamp_ > start_time_stamp) {
// Number of channels should be the same for both stand-alone and
// ACM-decoding.
EXPECT_EQ(audio_frame.num_channels_, channels);
}
decoded_samples = 0;
}
if (in_file_mono_.EndOfFile()) {
in_file_mono_.Rewind();
}
if (in_file_stereo_.EndOfFile()) {
in_file_stereo_.Rewind();
}
// Reset in case we ended with a lost packet.
channel->set_lost_packet(false);
}
void OpusTest::OpenOutFile(int test_number) {
std::string file_name;
std::stringstream file_stream;
file_stream << webrtc::test::OutputPath() << "opustest_out_"
<< test_number << ".pcm";
file_name = file_stream.str();
out_file_.Open(file_name, 48000, "wb");
file_stream.str("");
file_name = file_stream.str();
file_stream << webrtc::test::OutputPath() << "opusstandalone_out_"
<< test_number << ".pcm";
file_name = file_stream.str();
out_file_standalone_.Open(file_name, 48000, "wb");
}
} // namespace webrtc