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}diff --git a/modules/audio_coding/acm2/acm_receive_test.h b/modules/audio_coding/acm2/acm_receive_test.h
index 9d004c6..fce3d6b 100644
--- a/modules/audio_coding/acm2/acm_receive_test.h
+++ b/modules/audio_coding/acm2/acm_receive_test.h
@@ -33,8 +33,7 @@
enum NumOutputChannels : size_t {
kArbitraryChannels = 0,
kMonoOutput = 1,
- kStereoOutput = 2,
- kQuadOutput = 4
+ kStereoOutput = 2
};
AcmReceiveTestOldApi(PacketSource* packet_source,
diff --git a/modules/audio_coding/acm2/acm_send_test.cc b/modules/audio_coding/acm2/acm_send_test.cc
index b6110b6..98d673f 100644
--- a/modules/audio_coding/acm2/acm_send_test.cc
+++ b/modules/audio_coding/acm2/acm_send_test.cc
@@ -106,9 +106,13 @@
// 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_,
- input_frame_.mutable_data()));
+ 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_);
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index 1547b37..67ef556 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -477,9 +477,7 @@
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;
}
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index 13b9581..c9a03a1 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/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 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 @@
// 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 @@
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 @@
->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 @@
// 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 @@
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 @@
"_" + ::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 @@
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 @@
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(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 @@
// 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 @@
&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",