36b1a5fcec
const int16_t* data() const; int16_t* mutable_data(); - data() returns a zeroed static buffer on muted frames (to avoid unnecessary zeroing of the member buffer) and directly returns AudioFrame::data_ on unmuted frames. - mutable_data(), lazily zeroes AudioFrame::data_ if the frame is currently muted, sets muted=false, and returns AudioFrame::data_. These accessors serve to "force" callers to be aware of the mute state field, i.e. lazy zeroing is not the primary motivation. This change only optimizes handling of muted frames where it is somewhat trivial to do so. Other improvements requiring more significant structural changes will come later. BUG=webrtc:7343 TBR=henrika Review-Url: https://codereview.webrtc.org/2750783004 Cr-Commit-Position: refs/heads/master@{#18543}
391 lines
12 KiB
C++
391 lines
12 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_coding/test/opus_test.h"
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#include <assert.h>
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#include <string>
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#include "webrtc/common_types.h"
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#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
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#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
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#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
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#include "webrtc/modules/audio_coding/test/TestStereo.h"
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#include "webrtc/modules/audio_coding/test/utility.h"
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#include "webrtc/test/gtest.h"
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#include "webrtc/test/testsupport/fileutils.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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OpusTest::OpusTest()
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: acm_receiver_(AudioCodingModule::Create(0)),
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channel_a2b_(NULL),
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counter_(0),
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payload_type_(255),
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rtp_timestamp_(0) {}
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OpusTest::~OpusTest() {
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if (channel_a2b_ != NULL) {
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delete channel_a2b_;
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channel_a2b_ = NULL;
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}
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if (opus_mono_encoder_ != NULL) {
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WebRtcOpus_EncoderFree(opus_mono_encoder_);
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opus_mono_encoder_ = NULL;
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}
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if (opus_stereo_encoder_ != NULL) {
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WebRtcOpus_EncoderFree(opus_stereo_encoder_);
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opus_stereo_encoder_ = NULL;
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}
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if (opus_mono_decoder_ != NULL) {
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WebRtcOpus_DecoderFree(opus_mono_decoder_);
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opus_mono_decoder_ = NULL;
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}
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if (opus_stereo_decoder_ != NULL) {
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WebRtcOpus_DecoderFree(opus_stereo_decoder_);
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opus_stereo_decoder_ = NULL;
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}
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}
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void OpusTest::Perform() {
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#ifndef WEBRTC_CODEC_OPUS
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// Opus isn't defined, exit.
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return;
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#else
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uint16_t frequency_hz;
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size_t audio_channels;
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int16_t test_cntr = 0;
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// Open both mono and stereo test files in 32 kHz.
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const std::string file_name_stereo =
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webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm");
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const std::string file_name_mono =
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webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
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frequency_hz = 32000;
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in_file_stereo_.Open(file_name_stereo, frequency_hz, "rb");
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in_file_stereo_.ReadStereo(true);
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in_file_mono_.Open(file_name_mono, frequency_hz, "rb");
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in_file_mono_.ReadStereo(false);
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// Create Opus encoders for mono and stereo.
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ASSERT_GT(WebRtcOpus_EncoderCreate(&opus_mono_encoder_, 1, 0), -1);
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ASSERT_GT(WebRtcOpus_EncoderCreate(&opus_stereo_encoder_, 2, 1), -1);
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// Create Opus decoders for mono and stereo for stand-alone testing of Opus.
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ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_mono_decoder_, 1), -1);
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ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_stereo_decoder_, 2), -1);
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WebRtcOpus_DecoderInit(opus_mono_decoder_);
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WebRtcOpus_DecoderInit(opus_stereo_decoder_);
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ASSERT_TRUE(acm_receiver_.get() != NULL);
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EXPECT_EQ(0, acm_receiver_->InitializeReceiver());
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// Register Opus stereo as receiving codec.
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CodecInst opus_codec_param;
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int codec_id = acm_receiver_->Codec("opus", 48000, 2);
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EXPECT_EQ(0, acm_receiver_->Codec(codec_id, &opus_codec_param));
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payload_type_ = opus_codec_param.pltype;
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EXPECT_EQ(true,
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acm_receiver_->RegisterReceiveCodec(
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opus_codec_param.pltype, CodecInstToSdp(opus_codec_param)));
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// Create and connect the channel.
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channel_a2b_ = new TestPackStereo;
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channel_a2b_->RegisterReceiverACM(acm_receiver_.get());
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//
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// Test Stereo.
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//
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channel_a2b_->set_codec_mode(kStereo);
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audio_channels = 2;
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test_cntr++;
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OpenOutFile(test_cntr);
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// Run Opus with 2.5 ms frame size.
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Run(channel_a2b_, audio_channels, 64000, 120);
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// Run Opus with 5 ms frame size.
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Run(channel_a2b_, audio_channels, 64000, 240);
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// Run Opus with 10 ms frame size.
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Run(channel_a2b_, audio_channels, 64000, 480);
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// Run Opus with 20 ms frame size.
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Run(channel_a2b_, audio_channels, 64000, 960);
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// Run Opus with 40 ms frame size.
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Run(channel_a2b_, audio_channels, 64000, 1920);
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// Run Opus with 60 ms frame size.
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Run(channel_a2b_, audio_channels, 64000, 2880);
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out_file_.Close();
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out_file_standalone_.Close();
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//
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// Test Opus stereo with packet-losses.
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//
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test_cntr++;
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OpenOutFile(test_cntr);
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// Run Opus with 20 ms frame size, 1% packet loss.
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Run(channel_a2b_, audio_channels, 64000, 960, 1);
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// Run Opus with 20 ms frame size, 5% packet loss.
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Run(channel_a2b_, audio_channels, 64000, 960, 5);
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// Run Opus with 20 ms frame size, 10% packet loss.
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Run(channel_a2b_, audio_channels, 64000, 960, 10);
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out_file_.Close();
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out_file_standalone_.Close();
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//
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// Test Mono.
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//
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channel_a2b_->set_codec_mode(kMono);
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audio_channels = 1;
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test_cntr++;
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OpenOutFile(test_cntr);
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// Register Opus mono as receiving codec.
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opus_codec_param.channels = 1;
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EXPECT_EQ(true,
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acm_receiver_->RegisterReceiveCodec(
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opus_codec_param.pltype, CodecInstToSdp(opus_codec_param)));
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// Run Opus with 2.5 ms frame size.
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Run(channel_a2b_, audio_channels, 32000, 120);
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// Run Opus with 5 ms frame size.
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Run(channel_a2b_, audio_channels, 32000, 240);
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// Run Opus with 10 ms frame size.
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Run(channel_a2b_, audio_channels, 32000, 480);
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// Run Opus with 20 ms frame size.
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Run(channel_a2b_, audio_channels, 32000, 960);
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// Run Opus with 40 ms frame size.
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Run(channel_a2b_, audio_channels, 32000, 1920);
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// Run Opus with 60 ms frame size.
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Run(channel_a2b_, audio_channels, 32000, 2880);
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out_file_.Close();
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out_file_standalone_.Close();
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//
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// Test Opus mono with packet-losses.
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//
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test_cntr++;
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OpenOutFile(test_cntr);
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// Run Opus with 20 ms frame size, 1% packet loss.
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Run(channel_a2b_, audio_channels, 64000, 960, 1);
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// Run Opus with 20 ms frame size, 5% packet loss.
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Run(channel_a2b_, audio_channels, 64000, 960, 5);
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// Run Opus with 20 ms frame size, 10% packet loss.
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Run(channel_a2b_, audio_channels, 64000, 960, 10);
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// Close the files.
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in_file_stereo_.Close();
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in_file_mono_.Close();
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out_file_.Close();
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out_file_standalone_.Close();
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#endif
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}
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void OpusTest::Run(TestPackStereo* channel, size_t channels, int bitrate,
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size_t frame_length, int percent_loss) {
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AudioFrame audio_frame;
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int32_t out_freq_hz_b = out_file_.SamplingFrequency();
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const size_t kBufferSizeSamples = 480 * 12 * 2; // 120 ms stereo audio.
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int16_t audio[kBufferSizeSamples];
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int16_t out_audio[kBufferSizeSamples];
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int16_t audio_type;
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size_t written_samples = 0;
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size_t read_samples = 0;
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size_t decoded_samples = 0;
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bool first_packet = true;
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uint32_t start_time_stamp = 0;
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channel->reset_payload_size();
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counter_ = 0;
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// Set encoder rate.
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EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_mono_encoder_, bitrate));
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EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_stereo_encoder_, bitrate));
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#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
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// If we are on Android, iOS and/or ARM, use a lower complexity setting as
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// default.
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const int kOpusComplexity5 = 5;
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EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_mono_encoder_, kOpusComplexity5));
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EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_stereo_encoder_,
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kOpusComplexity5));
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#endif
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// Fast-forward 1 second (100 blocks) since the files start with silence.
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in_file_stereo_.FastForward(100);
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in_file_mono_.FastForward(100);
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// Limit the runtime to 1000 blocks of 10 ms each.
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for (size_t audio_length = 0; audio_length < 1000; audio_length += 10) {
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bool lost_packet = false;
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// Get 10 msec of audio.
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if (channels == 1) {
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if (in_file_mono_.EndOfFile()) {
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break;
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}
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in_file_mono_.Read10MsData(audio_frame);
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} else {
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if (in_file_stereo_.EndOfFile()) {
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break;
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}
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in_file_stereo_.Read10MsData(audio_frame);
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}
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// If input audio is sampled at 32 kHz, resampling to 48 kHz is required.
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EXPECT_EQ(480,
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resampler_.Resample10Msec(audio_frame.data(),
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audio_frame.sample_rate_hz_,
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48000,
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channels,
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kBufferSizeSamples - written_samples,
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&audio[written_samples]));
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written_samples += 480 * channels;
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// Sometimes we need to loop over the audio vector to produce the right
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// number of packets.
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size_t loop_encode = (written_samples - read_samples) /
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(channels * frame_length);
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if (loop_encode > 0) {
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const size_t kMaxBytes = 1000; // Maximum number of bytes for one packet.
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size_t bitstream_len_byte;
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uint8_t bitstream[kMaxBytes];
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for (size_t i = 0; i < loop_encode; i++) {
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int bitstream_len_byte_int = WebRtcOpus_Encode(
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(channels == 1) ? opus_mono_encoder_ : opus_stereo_encoder_,
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&audio[read_samples], frame_length, kMaxBytes, bitstream);
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ASSERT_GE(bitstream_len_byte_int, 0);
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bitstream_len_byte = static_cast<size_t>(bitstream_len_byte_int);
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// Simulate packet loss by setting |packet_loss_| to "true" in
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// |percent_loss| percent of the loops.
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// TODO(tlegrand): Move handling of loss simulation to TestPackStereo.
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if (percent_loss > 0) {
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if (counter_ == floor((100 / percent_loss) + 0.5)) {
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counter_ = 0;
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lost_packet = true;
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channel->set_lost_packet(true);
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} else {
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lost_packet = false;
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channel->set_lost_packet(false);
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}
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counter_++;
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}
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// Run stand-alone Opus decoder, or decode PLC.
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if (channels == 1) {
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if (!lost_packet) {
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decoded_samples += WebRtcOpus_Decode(
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opus_mono_decoder_, bitstream, bitstream_len_byte,
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&out_audio[decoded_samples * channels], &audio_type);
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} else {
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decoded_samples += WebRtcOpus_DecodePlc(
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opus_mono_decoder_, &out_audio[decoded_samples * channels], 1);
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}
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} else {
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if (!lost_packet) {
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decoded_samples += WebRtcOpus_Decode(
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opus_stereo_decoder_, bitstream, bitstream_len_byte,
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&out_audio[decoded_samples * channels], &audio_type);
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} else {
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decoded_samples += WebRtcOpus_DecodePlc(
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opus_stereo_decoder_, &out_audio[decoded_samples * channels],
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1);
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}
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}
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// Send data to the channel. "channel" will handle the loss simulation.
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channel->SendData(kAudioFrameSpeech, payload_type_, rtp_timestamp_,
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bitstream, bitstream_len_byte, NULL);
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if (first_packet) {
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first_packet = false;
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start_time_stamp = rtp_timestamp_;
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}
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rtp_timestamp_ += static_cast<uint32_t>(frame_length);
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read_samples += frame_length * channels;
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}
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if (read_samples == written_samples) {
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read_samples = 0;
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written_samples = 0;
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}
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}
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// Run received side of ACM.
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bool muted;
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ASSERT_EQ(
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0, acm_receiver_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted));
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ASSERT_FALSE(muted);
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// Write output speech to file.
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out_file_.Write10MsData(
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audio_frame.data(),
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audio_frame.samples_per_channel_ * audio_frame.num_channels_);
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// Write stand-alone speech to file.
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out_file_standalone_.Write10MsData(out_audio, decoded_samples * channels);
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if (audio_frame.timestamp_ > start_time_stamp) {
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// Number of channels should be the same for both stand-alone and
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// ACM-decoding.
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EXPECT_EQ(audio_frame.num_channels_, channels);
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}
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decoded_samples = 0;
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}
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if (in_file_mono_.EndOfFile()) {
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in_file_mono_.Rewind();
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}
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if (in_file_stereo_.EndOfFile()) {
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in_file_stereo_.Rewind();
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}
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// Reset in case we ended with a lost packet.
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channel->set_lost_packet(false);
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}
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void OpusTest::OpenOutFile(int test_number) {
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std::string file_name;
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std::stringstream file_stream;
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file_stream << webrtc::test::OutputPath() << "opustest_out_"
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<< test_number << ".pcm";
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file_name = file_stream.str();
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out_file_.Open(file_name, 48000, "wb");
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file_stream.str("");
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file_name = file_stream.str();
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file_stream << webrtc::test::OutputPath() << "opusstandalone_out_"
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<< test_number << ".pcm";
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file_name = file_stream.str();
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out_file_standalone_.Open(file_name, 48000, "wb");
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}
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} // namespace webrtc
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