Revert "Reland of https://webrtc-review.googlesource.com/c/src/+/114883"

This reverts commit 5341aaccdb64e3336abf5875e8828222446adffa.

Reason for revert: Order of initialization of global static strings.

Original change's description:
> Reland of https://webrtc-review.googlesource.com/c/src/+/114883
> 
> The difference to the original is new bitexactness strings AND
> global static file string constants. The reason for reland is breaking
> downstream projects.
> 
> Original CL description:
> 
> Tests for multi-stream Opus.
> 
> This CL (mainly) adds bit-exactness tests for multi-stream Opus. The
> tests are in audio_coding_unittest.cc. Some refactoring of
> AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it
> possible. A few checks for "channels \in {1, 2}" are replaced with
> "channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few
> other changes are made to be able to write and read multi-channel WAV
> files.
> 
> The SDP changes are NOT included; as of this CL there is no way to set
> up a multi-stream opus en/de-coder from SDP strings.
> 
> Bug: webrtc:8649
> Change-Id: I9fd47c790c241c1876c4a731b0840bec30b4f1b2
> Reviewed-on: https://webrtc-review.googlesource.com/c/123387
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26774}

TBR=aleloi@webrtc.org,ossu@webrtc.org

Change-Id: I88060f2050ccee83d6091b042a10f79b3c4edc47
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8649
Reviewed-on: https://webrtc-review.googlesource.com/c/123580
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26777}

api/audio_codecs/opus/audio_encoder_opus_config.cc

@@ -55,8 +55,7 @@ AudioEncoderOpusConfig& AudioEncoderOpusConfig::operator=(
 bool AudioEncoderOpusConfig::IsOk() const {
   if (frame_size_ms <= 0 || frame_size_ms % 10 != 0)
     return false;
-  if (num_channels != 1 && num_channels != 2 && num_channels != 4 &&
-      num_channels != 6 && num_channels != 8)
+  if (num_channels != 1 && num_channels != 2)
     return false;
   if (!bitrate_bps)
     return false;

modules/BUILD.gn

@@ -126,7 +126,6 @@ if (rtc_include_tests) {
     "../resources/audio_coding/speech_mono_32_48kHz.pcm",
     "../resources/audio_coding/speech_4_channels_48k_one_second.wav",
     "../resources/audio_coding/testfile32kHz.pcm",
-    "../resources/audio_coding/testfile_fake_stereo_32kHz.pcm",
     "../resources/audio_coding/teststereo32kHz.pcm",
     "../resources/audio_device/audio_short16.pcm",
     "../resources/audio_device/audio_short44.pcm",

modules/audio_coding/acm2/acm_receive_test.h

@@ -33,8 +33,7 @@ class AcmReceiveTestOldApi {
   enum NumOutputChannels : size_t {
     kArbitraryChannels = 0,
     kMonoOutput = 1,
-    kStereoOutput = 2,
-    kQuadOutput = 4
+    kStereoOutput = 2
   };
 
   AcmReceiveTestOldApi(PacketSource* packet_source,

modules/audio_coding/acm2/acm_send_test.cc

@@ -106,9 +106,13 @@ std::unique_ptr<Packet> AcmSendTestOldApi::NextPacket() {
   // Insert audio and process until one packet is produced.
   while (clock_.TimeInMilliseconds() < test_duration_ms_) {
     clock_.AdvanceTimeMilliseconds(kBlockSizeMs);
-    RTC_CHECK(audio_source_->Read(
-        input_block_size_samples_ * input_frame_.num_channels_,
+    RTC_CHECK(audio_source_->Read(input_block_size_samples_,
                                   input_frame_.mutable_data()));
+    if (input_frame_.num_channels_ > 1) {
+      InputAudioFile::DuplicateInterleaved(
+          input_frame_.data(), input_block_size_samples_,
+          input_frame_.num_channels_, input_frame_.mutable_data());
+    }
     data_to_send_ = false;
     RTC_CHECK_GE(acm_->Add10MsData(input_frame_), 0);
     input_frame_.timestamp_ += static_cast<uint32_t>(input_block_size_samples_);

modules/audio_coding/acm2/audio_coding_module.cc

@@ -477,9 +477,7 @@ int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame,
     return -1;
   }
 
-  if (audio_frame.num_channels_ != 1 && audio_frame.num_channels_ != 2 &&
-      audio_frame.num_channels_ != 4 && audio_frame.num_channels_ != 6 &&
-      audio_frame.num_channels_ != 8) {
+  if (audio_frame.num_channels_ != 1 && audio_frame.num_channels_ != 2) {
     RTC_LOG(LS_ERROR) << "Cannot Add 10 ms audio, invalid number of channels.";
     return -1;
   }

modules/audio_coding/acm2/audio_coding_module_unittest.cc

@@ -17,7 +17,6 @@
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
-#include "api/audio_codecs/opus/audio_decoder_opus.h"
 #include "api/audio_codecs/opus/audio_encoder_opus.h"
 #include "modules/audio_coding/acm2/acm_receive_test.h"
 #include "modules/audio_coding/acm2/acm_send_test.h"
@@ -25,8 +24,6 @@
 #include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
 #include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
 #include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
-#include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
-#include "modules/audio_coding/codecs/opus/audio_encoder_opus.h"
 #include "modules/audio_coding/include/audio_coding_module.h"
 #include "modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "modules/audio_coding/neteq/tools/audio_checksum.h"
@@ -47,7 +44,6 @@
 #include "rtc_base/thread_annotations.h"
 #include "system_wrappers/include/clock.h"
 #include "system_wrappers/include/sleep.h"
-#include "test/audio_decoder_proxy_factory.h"
 #include "test/gtest.h"
 #include "test/mock_audio_decoder.h"
 #include "test/mock_audio_encoder.h"
@@ -66,16 +62,6 @@ const int kFrameSizeMs = 10;  // Multiple of 10.
 const int kFrameSizeSamples = kFrameSizeMs / 10 * kNumSamples10ms;
 const int kPayloadSizeBytes = kFrameSizeSamples * sizeof(int16_t);
 const uint8_t kPayloadType = 111;
-const std::string kTestFileMono32kHz =
-    webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
-const std::string kTestFileFakeStereo32kHz =
-    webrtc::test::ResourcePath("audio_coding/testfile_fake_stereo_32kHz",
-                               "pcm");
-const std::string kTestFileQuad48kHz =
-    webrtc::test::ResourcePath("audio_coding/speech_4_channels_48k_one_second",
-                               "wav");
-const std::string kTestFileSpeechMono16kHz =
-    webrtc::test::ResourcePath("audio_coding/speech_mono_16kHz", "pcm");
 }  // namespace
 
 class RtpUtility {
@@ -639,8 +625,9 @@ class AcmIsacMtTestOldApi : public AudioCodingModuleMtTestOldApi {
 
     // Set up input audio source to read from specified file, loop after 5
     // seconds, and deliver blocks of 10 ms.
-    audio_loop_.Init(kTestFileSpeechMono16kHz, 5 * kSampleRateHz,
-                     kNumSamples10ms);
+    const std::string input_file_name =
+        webrtc::test::ResourcePath("audio_coding/speech_mono_16kHz", "pcm");
+    audio_loop_.Init(input_file_name, 5 * kSampleRateHz, kNumSamples10ms);
 
     // Generate one packet to have something to insert.
     int loop_counter = 0;
@@ -754,8 +741,9 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
     AudioCodingModuleTestOldApi::SetUp();
     // Set up input audio source to read from specified file, loop after 5
     // seconds, and deliver blocks of 10 ms.
-    audio_loop_.Init(kTestFileSpeechMono16kHz, 5 * kSampleRateHz,
-                     kNumSamples10ms);
+    const std::string input_file_name =
+        webrtc::test::ResourcePath("audio_coding/speech_mono_16kHz", "pcm");
+    audio_loop_.Init(input_file_name, 5 * kSampleRateHz, kNumSamples10ms);
     RegisterCodec();  // Must be called before the threads start below.
     StartThreads();
   }
@@ -947,7 +935,7 @@ class AcmReceiverBitExactnessOldApi : public ::testing::Test {
             ->test_case_name() +
         "_" + ::testing::UnitTest::GetInstance()->current_test_info()->name() +
         "_output.wav";
-    test::OutputWavFile output_file(output_file_name, output_freq_hz, 1);
+    test::OutputWavFile output_file(output_file_name, output_freq_hz);
     test::AudioSinkFork output(&checksum, &output_file);
 
     test::AcmReceiveTestOldApi test(
@@ -1128,12 +1116,15 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test,
 
   // Sets up the test::AcmSendTest object. Returns true on success, otherwise
   // false.
-  bool SetUpSender(std::string input_file_name, int source_rate) {
+  bool SetUpSender() {
+    const std::string input_file_name =
+        webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
     // Note that |audio_source_| will loop forever. The test duration is set
     // explicitly by |kTestDurationMs|.
     audio_source_.reset(new test::InputAudioFile(input_file_name));
-    send_test_.reset(new test::AcmSendTestOldApi(audio_source_.get(),
-                                                 source_rate, kTestDurationMs));
+    static const int kSourceRateHz = 32000;
+    send_test_.reset(new test::AcmSendTestOldApi(
+        audio_source_.get(), kSourceRateHz, kTestDurationMs));
     return send_test_.get() != NULL;
   }
 
@@ -1166,11 +1157,7 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test,
   void Run(const std::string& audio_checksum_ref,
            const std::string& payload_checksum_ref,
            int expected_packets,
-           test::AcmReceiveTestOldApi::NumOutputChannels expected_channels,
-           rtc::scoped_refptr<AudioDecoderFactory> decoder_factory = nullptr) {
-    if (!decoder_factory) {
-      decoder_factory = CreateBuiltinAudioDecoderFactory();
-    }
+           test::AcmReceiveTestOldApi::NumOutputChannels expected_channels) {
     // Set up the receiver used to decode the packets and verify the decoded
     // output.
     test::AudioChecksum audio_checksum;
@@ -1182,12 +1169,12 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test,
         "_" + ::testing::UnitTest::GetInstance()->current_test_info()->name() +
         "_output.wav";
     const int kOutputFreqHz = 8000;
-    test::OutputWavFile output_file(output_file_name, kOutputFreqHz,
-                                    expected_channels);
+    test::OutputWavFile output_file(output_file_name, kOutputFreqHz);
     // Have the output audio sent both to file and to the checksum calculator.
     test::AudioSinkFork output(&audio_checksum, &output_file);
     test::AcmReceiveTestOldApi receive_test(this, &output, kOutputFreqHz,
-                                            expected_channels, decoder_factory);
+                                            expected_channels,
+                                            CreateBuiltinAudioDecoderFactory());
     ASSERT_NO_FATAL_FAILURE(receive_test.RegisterDefaultCodecs());
 
     // This is where the actual test is executed.
@@ -1263,8 +1250,7 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test,
                  int payload_type,
                  int codec_frame_size_samples,
                  int codec_frame_size_rtp_timestamps) {
-    ASSERT_TRUE(SetUpSender(
-        channels == 1 ? kTestFileMono32kHz : kTestFileFakeStereo32kHz, 32000));
+    ASSERT_TRUE(SetUpSender());
     ASSERT_TRUE(RegisterSendCodec(codec_name, codec_sample_rate_hz, channels,
                                   payload_type, codec_frame_size_samples,
                                   codec_frame_size_rtp_timestamps));
@@ -1273,7 +1259,7 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test,
   void SetUpTestExternalEncoder(
       std::unique_ptr<AudioEncoder> external_speech_encoder,
       int payload_type) {
-    ASSERT_TRUE(send_test_);
+    ASSERT_TRUE(SetUpSender());
     RegisterExternalSendCodec(std::move(external_speech_encoder), payload_type);
   }
 
@@ -1495,59 +1481,17 @@ TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms) {
 TEST_F(AcmSenderBitExactnessNewApi, MAYBE_OpusFromFormat_stereo_20ms) {
   const auto config = AudioEncoderOpus::SdpToConfig(
       SdpAudioFormat("opus", 48000, 2, {{"stereo", "1"}}));
-  ASSERT_TRUE(SetUpSender(kTestFileFakeStereo32kHz, 32000));
   ASSERT_NO_FATAL_FAILURE(SetUpTestExternalEncoder(
       AudioEncoderOpus::MakeAudioEncoder(*config, 120), 120));
   Run(audio_checksum, payload_checksum, 50,
       test::AcmReceiveTestOldApi::kStereoOutput);
 }
 
-TEST_F(AcmSenderBitExactnessNewApi, OpusManyChannels) {
-  constexpr int kNumChannels = 4;
-  constexpr int kOpusPayloadType = 120;
-  constexpr int kBitrateBps = 128000;
-
-  // Read a 4 channel file at 48kHz.
-  ASSERT_TRUE(SetUpSender(kTestFileQuad48kHz, 48000));
-
-  // TODO(webrtc:8649): change to higher level
-  // AudioEncoderOpus::MakeAudioEncoder once a multistream encoder can be set up
-  // from SDP.
-  AudioEncoderOpusConfig config = *AudioEncoderOpus::SdpToConfig(
-      SdpAudioFormat("opus", 48000, 2, {{"stereo", "1"}}));
-  config.num_channels = kNumChannels;
-  config.bitrate_bps = kBitrateBps;
-
-  ASSERT_NO_FATAL_FAILURE(SetUpTestExternalEncoder(
-      absl::make_unique<AudioEncoderOpusImpl>(config, kOpusPayloadType),
-      kOpusPayloadType));
-
-  AudioDecoderOpusImpl opus_decoder(kNumChannels);
-
-  rtc::scoped_refptr<AudioDecoderFactory> decoder_factory =
-      new rtc::RefCountedObject<test::AudioDecoderProxyFactory>(&opus_decoder);
-
-  // Set up an EXTERNAL DECODER to parse 4 channels.
-  Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(  // audio checksum
-          "b70470884d9a8613eff019b0d1c8876e|d0a73d377e0ca1be6b06e989e0ad2c35",
-          "d0a73d377e0ca1be6b06e989e0ad2c35",
-          "b45d2ce5fc4723e9eb41350af9c68f56", "android arm64 audio checksum",
-          "1c9a3c9dacdd4b8fc9ff608227e531f2"),
-      // payload_checksum,
-      AcmReceiverBitExactnessOldApi::PlatformChecksum(  // payload checksum
-          "c2e7d40f8269ef754bd86d6be9623fa7|76de0f4992e3937ca60d35bbb0d308d6",
-          "76de0f4992e3937ca60d35bbb0d308d6",
-          "2a310aca965c16c2dfd61a9f9fc0c877", "android arm64 payload checksum",
-          "2294f4b61fb8f174f5196776a0a49be7"),
-      50, test::AcmReceiveTestOldApi::kQuadOutput, decoder_factory);
-}
-
 TEST_F(AcmSenderBitExactnessNewApi, OpusFromFormat_stereo_20ms_voip) {
   auto config = AudioEncoderOpus::SdpToConfig(
       SdpAudioFormat("opus", 48000, 2, {{"stereo", "1"}}));
   // If not set, default will be kAudio in case of stereo.
   config->application = AudioEncoderOpusConfig::ApplicationMode::kVoip;
-  ASSERT_TRUE(SetUpSender(kTestFileFakeStereo32kHz, 32000));
   ASSERT_NO_FATAL_FAILURE(SetUpTestExternalEncoder(
       AudioEncoderOpus::MakeAudioEncoder(*config, 120), 120));
   // Checksum depends on libopus being compiled with or without SSE.
@@ -1578,9 +1522,11 @@ class AcmSetBitRateTest : public ::testing::Test {
   // Sets up the test::AcmSendTest object. Returns true on success, otherwise
   // false.
   bool SetUpSender() {
+    const std::string input_file_name =
+        webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
     // Note that |audio_source_| will loop forever. The test duration is set
     // explicitly by |kTestDurationMs|.
-    audio_source_.reset(new test::InputAudioFile(kTestFileMono32kHz));
+    audio_source_.reset(new test::InputAudioFile(input_file_name));
     static const int kSourceRateHz = 32000;
     send_test_.reset(new test::AcmSendTestOldApi(
         audio_source_.get(), kSourceRateHz, kTestDurationMs));
@@ -1829,7 +1775,6 @@ TEST_F(AcmSenderBitExactnessOldApi, External_Pcmu_20ms) {
           &encoder, static_cast<AudioEncoder::EncodedInfo (AudioEncoder::*)(
                         uint32_t, rtc::ArrayView<const int16_t>, rtc::Buffer*)>(
                         &AudioEncoderPcmU::Encode)));
-  ASSERT_TRUE(SetUpSender(kTestFileMono32kHz, 32000));
   ASSERT_NO_FATAL_FAILURE(
       SetUpTestExternalEncoder(std::move(mock_encoder), config.payload_type));
   Run("81a9d4c0bb72e9becc43aef124c981e9", "8f9b8750bd80fe26b6cbf6659b89f0f9",

modules/audio_coding/codecs/opus/audio_decoder_opus.cc

@@ -71,8 +71,7 @@ class OpusFrame : public AudioDecoder::EncodedAudioFrame {
 
 AudioDecoderOpusImpl::AudioDecoderOpusImpl(size_t num_channels)
     : channels_(num_channels) {
-  RTC_DCHECK(num_channels == 1 || num_channels == 2 || num_channels == 4 ||
-             num_channels == 6 || num_channels == 8);
+  RTC_DCHECK(num_channels == 1 || num_channels == 2);
   const int error = WebRtcOpus_DecoderCreate(&dec_state_, channels_);
   RTC_DCHECK(error == 0);
   WebRtcOpus_DecoderInit(dec_state_);

modules/audio_coding/neteq/tools/input_audio_file.cc

@@ -18,11 +18,9 @@ namespace test {
 InputAudioFile::InputAudioFile(const std::string file_name, bool loop_at_end)
     : loop_at_end_(loop_at_end) {
   fp_ = fopen(file_name.c_str(), "rb");
-  RTC_DCHECK(fp_) << file_name << " could not be opened.";
 }
 
 InputAudioFile::~InputAudioFile() {
-  RTC_DCHECK(fp_);
   fclose(fp_);
 }
 

modules/audio_coding/neteq/tools/output_wav_file.h

@@ -24,10 +24,8 @@ class OutputWavFile : public AudioSink {
  public:
   // Creates an OutputWavFile, opening a file named |file_name| for writing.
   // The output file is a PCM encoded wav file.
-  OutputWavFile(const std::string& file_name,
-                int sample_rate_hz,
-                int num_channels = 1)
-      : wav_writer_(file_name, sample_rate_hz, num_channels) {}
+  OutputWavFile(const std::string& file_name, int sample_rate_hz)
+      : wav_writer_(file_name, sample_rate_hz, 1) {}
 
   bool WriteArray(const int16_t* audio, size_t num_samples) override {
     wav_writer_.WriteSamples(audio, num_samples);

resources/audio_coding/testfile_fake_stereo_32kHz.pcm.sha1

@@ -1 +0,0 @@
-4f382602b5605dbbbf78451810ce644788681262