Add send-side bit-exactness test for AudioCoding Module
This test verifies bit exactness for the send-side of ACM. The test
setup is a chain of three different test classes:
test::AcmSendTest -> AcmSenderBitExactness -> test::AcmReceiveTest
The receiver side is driving the test by requesting new packets from
AcmSenderBitExactness::NextPacket(). This method, in turn, asks for the
packet from test::AcmSendTest::NextPacket, which inserts audio from the
input file until one packet is produced. (The input file loops
indefinitely.) Before passing the packet to the receiver, the
AcmSenderBitExactness class verifies the packet header and updates a
payload checksum with the new payload. The decoded output from the
receiver is also verified with a (separate) checksum.
The current CL only adds tests for 30 ms and 60 ms iSAC. More codecs
will be added in coming changes.
BUG=3521
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6949 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/modules/audio_coding/main/acm2/acm_send_test.cc b/modules/audio_coding/main/acm2/acm_send_test.cc
new file mode 100644
index 0000000..67cc9b2
--- /dev/null
+++ b/modules/audio_coding/main/acm2/acm_send_test.cc
@@ -0,0 +1,139 @@
+/*
+ * Copyright (c) 2014 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/acm2/acm_send_test.h"
+
+#include <assert.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
+
+namespace webrtc {
+namespace test {
+
+AcmSendTest::AcmSendTest(InputAudioFile* audio_source,
+ int source_rate_hz,
+ int test_duration_ms)
+ : clock_(0),
+ acm_(webrtc::AudioCodingModule::Create(0, &clock_)),
+ audio_source_(audio_source),
+ source_rate_hz_(source_rate_hz),
+ input_block_size_samples_(source_rate_hz_ * kBlockSizeMs / 1000),
+ codec_registered_(false),
+ test_duration_ms_(test_duration_ms),
+ frame_type_(kAudioFrameSpeech),
+ payload_type_(0),
+ timestamp_(0),
+ sequence_number_(0) {
+ input_frame_.sample_rate_hz_ = source_rate_hz_;
+ input_frame_.num_channels_ = 1;
+ input_frame_.samples_per_channel_ = input_block_size_samples_;
+ assert(input_block_size_samples_ * input_frame_.num_channels_ <=
+ AudioFrame::kMaxDataSizeSamples);
+ acm_->RegisterTransportCallback(this);
+}
+
+bool AcmSendTest::RegisterCodec(const char* payload_name,
+ int sampling_freq_hz,
+ int channels,
+ int payload_type,
+ int frame_size_samples) {
+ FATAL_ERROR_IF(AudioCodingModule::Codec(
+ payload_name, &codec_, sampling_freq_hz, channels) != 0);
+ codec_.pltype = payload_type;
+ codec_.pacsize = frame_size_samples;
+ codec_registered_ = (acm_->RegisterSendCodec(codec_) == 0);
+ assert(channels == 1); // TODO(henrik.lundin) Add multi-channel support.
+ input_frame_.num_channels_ = channels;
+ assert(input_block_size_samples_ * input_frame_.num_channels_ <=
+ AudioFrame::kMaxDataSizeSamples);
+ return codec_registered_;
+}
+
+Packet* AcmSendTest::NextPacket() {
+ assert(codec_registered_);
+ if (filter_.test(payload_type_)) {
+ // This payload type should be filtered out. Since the payload type is the
+ // same throughout the whole test run, no packet at all will be delivered.
+ // We can just as well signal that the test is over by returning NULL.
+ return NULL;
+ }
+ // Insert audio and process until one packet is produced.
+ while (clock_.TimeInMilliseconds() < test_duration_ms_) {
+ clock_.AdvanceTimeMilliseconds(kBlockSizeMs);
+ FATAL_ERROR_IF(
+ !audio_source_->Read(input_block_size_samples_, input_frame_.data_));
+ FATAL_ERROR_IF(acm_->Add10MsData(input_frame_) != 0);
+ input_frame_.timestamp_ += input_block_size_samples_;
+ int32_t encoded_bytes = acm_->Process();
+ if (encoded_bytes > 0) {
+ // Encoded packet received.
+ return CreatePacket();
+ }
+ }
+ // Test ended.
+ return NULL;
+}
+
+// This method receives the callback from ACM when a new packet is produced.
+int32_t AcmSendTest::SendData(FrameType frame_type,
+ uint8_t payload_type,
+ uint32_t timestamp,
+ const uint8_t* payload_data,
+ uint16_t payload_len_bytes,
+ const RTPFragmentationHeader* fragmentation) {
+ // Store the packet locally.
+ frame_type_ = frame_type;
+ payload_type_ = payload_type;
+ timestamp_ = timestamp;
+ last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes);
+ assert(last_payload_vec_.size() == payload_len_bytes);
+ return 0;
+}
+
+Packet* AcmSendTest::CreatePacket() {
+ const size_t kRtpHeaderSize = 12;
+ size_t allocated_bytes = last_payload_vec_.size() + kRtpHeaderSize;
+ uint8_t* packet_memory = new uint8_t[allocated_bytes];
+ // Populate the header bytes.
+ packet_memory[0] = 0x80;
+ packet_memory[1] = payload_type_;
+ packet_memory[2] = (sequence_number_ >> 8) & 0xFF;
+ packet_memory[3] = (sequence_number_) & 0xFF;
+ packet_memory[4] = (timestamp_ >> 24) & 0xFF;
+ packet_memory[5] = (timestamp_ >> 16) & 0xFF;
+ packet_memory[6] = (timestamp_ >> 8) & 0xFF;
+ packet_memory[7] = timestamp_ & 0xFF;
+ // Set SSRC to 0x12345678.
+ packet_memory[8] = 0x12;
+ packet_memory[9] = 0x34;
+ packet_memory[10] = 0x56;
+ packet_memory[11] = 0x78;
+
+ ++sequence_number_;
+
+ // Copy the payload data.
+ memcpy(packet_memory + kRtpHeaderSize,
+ &last_payload_vec_[0],
+ last_payload_vec_.size());
+ Packet* packet =
+ new Packet(packet_memory, allocated_bytes, clock_.TimeInMilliseconds());
+ assert(packet);
+ assert(packet->valid_header());
+ return packet;
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/modules/audio_coding/main/acm2/acm_send_test.h b/modules/audio_coding/main/acm2/acm_send_test.h
new file mode 100644
index 0000000..5e9bd97
--- /dev/null
+++ b/modules/audio_coding/main/acm2/acm_send_test.h
@@ -0,0 +1,85 @@
+/*
+ * Copyright (c) 2014 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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_
+
+#include <vector>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
+#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+
+namespace webrtc {
+
+namespace test {
+class InputAudioFile;
+class Packet;
+
+class AcmSendTest : public AudioPacketizationCallback, public PacketSource {
+ public:
+ AcmSendTest(InputAudioFile* audio_source,
+ int source_rate_hz,
+ int test_duration_ms);
+ virtual ~AcmSendTest() {}
+
+ // Registers the send codec. Returns true on success, false otherwise.
+ bool RegisterCodec(const char* payload_name,
+ int sampling_freq_hz,
+ int channels,
+ int payload_type,
+ int frame_size_samples);
+
+ // Returns the next encoded packet. Returns NULL if the test duration was
+ // exceeded. Ownership of the packet is handed over to the caller.
+ // Inherited from PacketSource.
+ Packet* NextPacket();
+
+ // Inherited from AudioPacketizationCallback.
+ virtual int32_t SendData(
+ FrameType frame_type,
+ uint8_t payload_type,
+ uint32_t timestamp,
+ const uint8_t* payload_data,
+ uint16_t payload_len_bytes,
+ const RTPFragmentationHeader* fragmentation) OVERRIDE;
+
+ private:
+ static const int kBlockSizeMs = 10;
+
+ // Creates a Packet object from the last packet produced by ACM (and received
+ // through the SendData method as a callback). Ownership of the new Packet
+ // object is transferred to the caller.
+ Packet* CreatePacket();
+
+ SimulatedClock clock_;
+ scoped_ptr<AudioCodingModule> acm_;
+ InputAudioFile* audio_source_;
+ int source_rate_hz_;
+ const int input_block_size_samples_;
+ AudioFrame input_frame_;
+ CodecInst codec_;
+ bool codec_registered_;
+ int test_duration_ms_;
+ // The following member variables are set whenever SendData() is called.
+ FrameType frame_type_;
+ int payload_type_;
+ uint32_t timestamp_;
+ uint16_t sequence_number_;
+ std::vector<uint8_t> last_payload_vec_;
+
+ DISALLOW_COPY_AND_ASSIGN(AcmSendTest);
+};
+
+} // namespace test
+} // namespace webrtc
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_
diff --git a/modules/audio_coding/main/acm2/audio_coding_module.gypi b/modules/audio_coding/main/acm2/audio_coding_module.gypi
index dccfe68..f88dbd3 100644
--- a/modules/audio_coding/main/acm2/audio_coding_module.gypi
+++ b/modules/audio_coding/main/acm2/audio_coding_module.gypi
@@ -130,6 +130,19 @@
],
}, # acm_receive_test
{
+ 'target_name': 'acm_send_test',
+ 'type': 'static_library',
+ 'dependencies': [
+ 'audio_coding_module',
+ 'neteq_unittest_tools',
+ '<(DEPTH)/testing/gtest.gyp:gtest',
+ ],
+ 'sources': [
+ 'acm_send_test.cc',
+ 'acm_send_test.h',
+ ],
+ }, # acm_send_test
+ {
'target_name': 'delay_test',
'type': 'executable',
'dependencies': [
diff --git a/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
index 7e2ede5..9c21fec 100644
--- a/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
@@ -12,12 +12,16 @@
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/md5digest.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_receive_test.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_send_test.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
#include "webrtc/modules/audio_coding/neteq/tools/audio_checksum.h"
#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
+#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
#include "webrtc/modules/audio_coding/neteq/tools/output_audio_file.h"
+#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/system_wrappers/interface/clock.h"
@@ -516,6 +520,19 @@
}
class AcmReceiverBitExactness : public ::testing::Test {
+ public:
+ static std::string PlatformChecksum(std::string win64,
+ std::string android,
+ std::string others) {
+#if defined(_WIN32) && defined(WEBRTC_ARCH_64_BITS)
+ return win64;
+#elif defined(WEBRTC_ANDROID)
+ return android;
+#else
+ return others;
+#endif
+ }
+
protected:
void Run(int output_freq_hz, const std::string& checksum_ref) {
const std::string input_file_name =
@@ -546,18 +563,6 @@
std::string checksum_string = checksum.Finish();
EXPECT_EQ(checksum_ref, checksum_string);
}
-
- static std::string PlatformChecksum(std::string win64,
- std::string android,
- std::string others) {
-#if defined(_WIN32) && defined(WEBRTC_ARCH_64_BITS)
- return win64;
-#elif defined(WEBRTC_ANDROID)
- return android;
-#else
- return others;
-#endif
- }
};
TEST_F(AcmReceiverBitExactness, 8kHzOutput) {
@@ -587,4 +592,188 @@
"76b9e99e0a3998aa28355e7a2bd836f7",
"89b4b19bdb4de40f1d88302ef8cb9f9b"));
}
+
+// This test verifies bit exactness for the send-side of ACM. The test setup is
+// a chain of three different test classes:
+//
+// test::AcmSendTest -> AcmSenderBitExactness -> test::AcmReceiveTest
+//
+// The receiver side is driving the test by requesting new packets from
+// AcmSenderBitExactness::NextPacket(). This method, in turn, asks for the
+// packet from test::AcmSendTest::NextPacket, which inserts audio from the
+// input file until one packet is produced. (The input file loops indefinitely.)
+// Before passing the packet to the receiver, this test class verifies the
+// packet header and updates a payload checksum with the new payload. The
+// decoded output from the receiver is also verified with a (separate) checksum.
+class AcmSenderBitExactness : public ::testing::Test,
+ public test::PacketSource {
+ protected:
+ static const int kTestDurationMs = 1000;
+
+ AcmSenderBitExactness()
+ : frame_size_rtp_timestamps_(0),
+ packet_count_(0),
+ payload_type_(0),
+ last_sequence_number_(0),
+ last_timestamp_(0) {}
+
+ // 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(input_file_name));
+ static const int kSourceRateHz = 32000;
+ send_test_.reset(new test::AcmSendTest(
+ audio_source_.get(), kSourceRateHz, kTestDurationMs));
+ return send_test_.get() != NULL;
+ }
+
+ // Registers a send codec in the test::AcmSendTest object. Returns true on
+ // success, false on failure.
+ bool RegisterSendCodec(const char* payload_name,
+ int sampling_freq_hz,
+ int channels,
+ int payload_type,
+ int frame_size_samples,
+ int frame_size_rtp_timestamps) {
+ payload_type_ = payload_type;
+ frame_size_rtp_timestamps_ = frame_size_rtp_timestamps;
+ return send_test_->RegisterCodec(payload_name,
+ sampling_freq_hz,
+ channels,
+ payload_type,
+ frame_size_samples);
+ }
+
+ // Runs the test. SetUpSender() and RegisterSendCodec() must have been called
+ // before calling this method.
+ void Run(const std::string& audio_checksum_ref,
+ const std::string& payload_checksum_ref,
+ int expected_packets) {
+ // Set up the receiver used to decode the packets and verify the decoded
+ // output.
+ test::AudioChecksum audio_checksum;
+ const std::string output_file_name =
+ webrtc::test::OutputPath() +
+ ::testing::UnitTest::GetInstance()
+ ->current_test_info()
+ ->test_case_name() +
+ "_" +
+ ::testing::UnitTest::GetInstance()->current_test_info()->name() +
+ "_output.pcm";
+ test::OutputAudioFile output_file(output_file_name);
+ // Have the output audio sent both to file and to the checksum calculator.
+ test::AudioSinkFork output(&audio_checksum, &output_file);
+ const int kOutputFreqHz = 8000;
+ test::AcmReceiveTest receive_test(this, &output, kOutputFreqHz);
+ ASSERT_NO_FATAL_FAILURE(receive_test.RegisterDefaultCodecs());
+
+ // This is where the actual test is executed.
+ receive_test.Run();
+
+ // Extract and verify the audio checksum.
+ std::string checksum_string = audio_checksum.Finish();
+ EXPECT_EQ(audio_checksum_ref, checksum_string);
+
+ // Extract and verify the payload checksum.
+ char checksum_result[rtc::Md5Digest::kSize];
+ payload_checksum_.Finish(checksum_result, rtc::Md5Digest::kSize);
+ checksum_string = rtc::hex_encode(checksum_result, rtc::Md5Digest::kSize);
+ EXPECT_EQ(payload_checksum_ref, checksum_string);
+
+ // Verify number of packets produced.
+ EXPECT_EQ(expected_packets, packet_count_);
+ }
+
+ // Returns a pointer to the next packet. Returns NULL if the source is
+ // depleted (i.e., the test duration is exceeded), or if an error occurred.
+ // Inherited from test::PacketSource.
+ test::Packet* NextPacket() OVERRIDE {
+ // Get the next packet from AcmSendTest. Ownership of |packet| is
+ // transferred to this method.
+ test::Packet* packet = send_test_->NextPacket();
+ if (!packet)
+ return NULL;
+
+ VerifyPacket(packet);
+ // TODO(henrik.lundin) Save the packet to file as well.
+
+ // Pass it on to the caller. The caller becomes the owner of |packet|.
+ return packet;
+ }
+
+ // Verifies the packet.
+ void VerifyPacket(const test::Packet* packet) {
+ EXPECT_TRUE(packet->valid_header());
+ // (We can check the header fields even if valid_header() is false.)
+ EXPECT_EQ(payload_type_, packet->header().payloadType);
+ if (packet_count_ > 0) {
+ // This is not the first packet.
+ uint16_t sequence_number_diff =
+ packet->header().sequenceNumber - last_sequence_number_;
+ EXPECT_EQ(1, sequence_number_diff);
+ uint32_t timestamp_diff = packet->header().timestamp - last_timestamp_;
+ EXPECT_EQ(frame_size_rtp_timestamps_, timestamp_diff);
+ }
+ ++packet_count_;
+ last_sequence_number_ = packet->header().sequenceNumber;
+ last_timestamp_ = packet->header().timestamp;
+ // Update the checksum.
+ payload_checksum_.Update(packet->payload(), packet->payload_length_bytes());
+ }
+
+ void SetUpTest(const char* codec_name,
+ int codec_sample_rate_hz,
+ int channels,
+ int payload_type,
+ int codec_frame_size_samples,
+ int codec_frame_size_rtp_timestamps) {
+ ASSERT_TRUE(SetUpSender());
+ ASSERT_TRUE(RegisterSendCodec(codec_name,
+ codec_sample_rate_hz,
+ channels,
+ payload_type,
+ codec_frame_size_samples,
+ codec_frame_size_rtp_timestamps));
+ }
+
+ scoped_ptr<test::AcmSendTest> send_test_;
+ scoped_ptr<test::InputAudioFile> audio_source_;
+ uint32_t frame_size_rtp_timestamps_;
+ int packet_count_;
+ uint8_t payload_type_;
+ uint16_t last_sequence_number_;
+ uint32_t last_timestamp_;
+ rtc::Md5Digest payload_checksum_;
+};
+
+TEST_F(AcmSenderBitExactness, IsacWb30ms) {
+ ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 480, 480));
+ Run(AcmReceiverBitExactness::PlatformChecksum(
+ "c7e5bdadfa2871df95639fcc297cf23d",
+ "0499ca260390769b3172136faad925b9",
+ "0b58f9eeee43d5891f5f6c75e77984a3"),
+ AcmReceiverBitExactness::PlatformChecksum(
+ "d42cb5195463da26c8129bbfe73a22e6",
+ "83de248aea9c3c2bd680b6952401b4ca",
+ "3c79f16f34218271f3dca4e2b1dfe1bb"),
+ 33);
+}
+
+TEST_F(AcmSenderBitExactness, IsacWb60ms) {
+ ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 960, 960));
+ Run(AcmReceiverBitExactness::PlatformChecksum(
+ "14d63c5f08127d280e722e3191b73bdd",
+ "8da003e16c5371af2dc2be79a50f9076",
+ "1ad29139a04782a33daad8c2b9b35875"),
+ AcmReceiverBitExactness::PlatformChecksum(
+ "ebe04a819d3a9d83a83a17f271e1139a",
+ "97aeef98553b5a4b5a68f8b716e8eaf0",
+ "9e0a0ab743ad987b55b8e14802769c56"),
+ 16);
+}
+
} // namespace webrtc
diff --git a/modules/modules.gyp b/modules/modules.gyp
index e6919f1..f68ea7c 100644
--- a/modules/modules.gyp
+++ b/modules/modules.gyp
@@ -69,6 +69,7 @@
],
'dependencies': [
'acm_receive_test',
+ 'acm_send_test',
'audio_coding_module',
'audio_processing',
'bitrate_controller',
