audio_coding: remove "main" directory

webrtc/modules/audio_coding/main/test/opus_test.cc

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