Index: webrtc/modules/audio_coding/main/test/opus_test.cc |
diff --git a/webrtc/modules/audio_coding/main/test/opus_test.cc b/webrtc/modules/audio_coding/main/test/opus_test.cc |
deleted file mode 100644 |
index 27cc40aa3c74e1ddabd8ac96e3c9559b4d2ed005..0000000000000000000000000000000000000000 |
--- a/webrtc/modules/audio_coding/main/test/opus_test.cc |
+++ /dev/null |
@@ -1,380 +0,0 @@ |
-/* |
- * 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/main/test/opus_test.h" |
- |
-#include <assert.h> |
- |
-#include <string> |
- |
-#include "testing/gtest/include/gtest/gtest.h" |
-#include "webrtc/common_types.h" |
-#include "webrtc/engine_configurations.h" |
-#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h" |
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h" |
-#include "webrtc/modules/audio_coding/main/test/TestStereo.h" |
-#include "webrtc/modules/audio_coding/main/test/utility.h" |
-#include "webrtc/system_wrappers/include/trace.h" |
-#include "webrtc/test/testsupport/fileutils.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; |
- int 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(0, acm_receiver_->RegisterReceiveCodec(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(0, acm_receiver_->RegisterReceiveCodec(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, int channels, int bitrate, |
- int frame_length, int percent_loss) { |
- AudioFrame audio_frame; |
- int32_t out_freq_hz_b = out_file_.SamplingFrequency(); |
- const int kBufferSizeSamples = 480 * 12 * 2; // Can hold 120 ms stereo audio. |
- int16_t audio[kBufferSizeSamples]; |
- int16_t out_audio[kBufferSizeSamples]; |
- int16_t audio_type; |
- int written_samples = 0; |
- int read_samples = 0; |
- int 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 |
- |
- // Make sure the runtime is less than 60 seconds to pass Android test. |
- for (size_t audio_length = 0; audio_length < 10000; 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. |
- int loop_encode = (written_samples - read_samples) / |
- (channels * frame_length); |
- |
- if (loop_encode > 0) { |
- const int kMaxBytes = 1000; // Maximum number of bytes for one packet. |
- size_t bitstream_len_byte; |
- uint8_t bitstream[kMaxBytes]; |
- for (int 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_ += frame_length; |
- read_samples += frame_length * channels; |
- } |
- if (read_samples == written_samples) { |
- read_samples = 0; |
- written_samples = 0; |
- } |
- } |
- |
- // Run received side of ACM. |
- ASSERT_EQ(0, acm_receiver_->PlayoutData10Ms(out_freq_hz_b, &audio_frame)); |
- |
- // 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, static_cast<size_t>(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 |