diff --git a/resources/utility/encapsulated_pcm16b_8khz.wav.sha1 b/resources/utility/encapsulated_pcm16b_8khz.wav.sha1 new file mode 100644 index 0000000000..480cfea43f --- /dev/null +++ b/resources/utility/encapsulated_pcm16b_8khz.wav.sha1 @@ -0,0 +1 @@ +43092df43f4093e474c41cd74e3085e7ca401c7d \ No newline at end of file diff --git a/resources/utility/encapsulated_pcmu_8khz.wav.sha1 b/resources/utility/encapsulated_pcmu_8khz.wav.sha1 new file mode 100644 index 0000000000..e2269f7e06 --- /dev/null +++ b/resources/utility/encapsulated_pcmu_8khz.wav.sha1 @@ -0,0 +1 @@ +d96caa3ff9c48559ec37784791887994d28f3b5a \ No newline at end of file diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc index ed4b086949..b8bb12ca48 100644 --- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc +++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc @@ -1835,6 +1835,7 @@ int AudioCodingModuleImpl::IncomingPayload(const uint8_t* incoming_payload, aux_rtp_header_->type.Audio.channel = 1; } + aux_rtp_header_->header.timestamp = timestamp; IncomingPacket(incoming_payload, payload_length, *aux_rtp_header_); // Get ready for the next payload. aux_rtp_header_->header.sequenceNumber++; diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp index 47e0a74b1b..10fa7ba6de 100644 --- a/webrtc/modules/modules.gyp +++ b/webrtc/modules/modules.gyp @@ -221,6 +221,7 @@ 'rtp_rtcp/test/testAPI/test_api_rtcp.cc', 'rtp_rtcp/test/testAPI/test_api_video.cc', 'utility/source/audio_frame_operations_unittest.cc', + 'utility/source/file_player_unittests.cc', 'video_coding/codecs/test/packet_manipulator_unittest.cc', 'video_coding/codecs/test/stats_unittest.cc', 'video_coding/codecs/test/videoprocessor_unittest.cc', diff --git a/webrtc/modules/modules_unittests.isolate b/webrtc/modules/modules_unittests.isolate index 57d1307052..74b713f648 100644 --- a/webrtc/modules/modules_unittests.isolate +++ b/webrtc/modules/modules_unittests.isolate @@ -93,6 +93,8 @@ '../../resources/synthetic-trace.rx', '../../resources/tmobile-downlink.rx', '../../resources/tmobile-uplink.rx', + '../../resources/utility/encapsulated_pcm16b_8khz.wav', + '../../resources/utility/encapsulated_pcmu_8khz.wav', '../../resources/verizon3g-downlink.rx', '../../resources/verizon3g-uplink.rx', '../../resources/verizon4g-downlink.rx', diff --git a/webrtc/modules/utility/source/file_player_unittests.cc b/webrtc/modules/utility/source/file_player_unittests.cc new file mode 100644 index 0000000000..d430d9f59a --- /dev/null +++ b/webrtc/modules/utility/source/file_player_unittests.cc @@ -0,0 +1,106 @@ +/* + * 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. + */ + +// Unit tests for FilePlayer. + +#include "webrtc/modules/utility/interface/file_player.h" + +#include +#include + +#include "gflags/gflags.h" +#include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/base/md5digest.h" +#include "webrtc/base/stringencode.h" +#include "webrtc/test/testsupport/fileutils.h" + +DEFINE_bool(file_player_output, false, "Generate reference files."); + +namespace webrtc { + +class FilePlayerTest : public ::testing::Test { + protected: + static const uint32_t kId = 0; + static const FileFormats kFileFormat = kFileFormatWavFile; + static const int kSampleRateHz = 8000; + + FilePlayerTest() + : player_(FilePlayer::CreateFilePlayer(kId, kFileFormat)), + output_file_(NULL) {} + + virtual void SetUp() OVERRIDE { + if (FLAGS_file_player_output) { + std::string output_file = + webrtc::test::OutputPath() + "file_player_unittest_out.pcm"; + output_file_ = fopen(output_file.c_str(), "wb"); + ASSERT_TRUE(output_file_ != NULL); + } + } + + virtual void TearDown() OVERRIDE { + if (output_file_) + fclose(output_file_); + } + + ~FilePlayerTest() { FilePlayer::DestroyFilePlayer(player_); } + + void PlayFileAndCheck(const std::string& input_file, + const std::string& ref_checksum, + int output_length_ms) { + const float kScaling = 1; + ASSERT_EQ(0, + player_->StartPlayingFile( + input_file.c_str(), false, 0, kScaling, 0, 0, NULL)); + rtc::Md5Digest checksum; + for (int i = 0; i < output_length_ms / 10; ++i) { + int16_t out[10 * kSampleRateHz / 1000] = {0}; + int num_samples; + EXPECT_EQ(0, + player_->Get10msAudioFromFile(out, num_samples, kSampleRateHz)); + checksum.Update(out, num_samples * sizeof(out[0])); + if (FLAGS_file_player_output) { + ASSERT_EQ(static_cast(num_samples), + fwrite(out, sizeof(out[0]), num_samples, output_file_)); + } + } + char checksum_result[rtc::Md5Digest::kSize]; + EXPECT_EQ(rtc::Md5Digest::kSize, + checksum.Finish(checksum_result, rtc::Md5Digest::kSize)); + EXPECT_EQ(ref_checksum, + rtc::hex_encode(checksum_result, sizeof(checksum_result))); + } + + FilePlayer* player_; + FILE* output_file_; +}; + +TEST_F(FilePlayerTest, PlayWavPcmuFile) { + const std::string kFileName = + test::ResourcePath("utility/encapsulated_pcmu_8khz", "wav"); + // The file is longer than this, but keeping the output shorter limits the + // runtime for the test. + const int kOutputLengthMs = 10000; + const std::string kRefChecksum = "c74e7fd432d439b1311e1c16815b3e9a"; + + PlayFileAndCheck(kFileName, kRefChecksum, kOutputLengthMs); +} + +TEST_F(FilePlayerTest, PlayWavPcm16File) { + const std::string kFileName = + test::ResourcePath("utility/encapsulated_pcm16b_8khz", "wav"); + // The file is longer than this, but keeping the output shorter limits the + // runtime for the test. + const int kOutputLengthMs = 10000; + const std::string kRefChecksum = "e41d7e1dac8aeae9f21e8e03cd7ecd71"; + + PlayFileAndCheck(kFileName, kRefChecksum, kOutputLengthMs); +} + +} // namespace webrtc