Reduce the runtime of some ACM tests in modules_tests
By reducing the length of the audio input, the total runtime of
$ out/Debug/modules_tests --gtest_filter=AudioCodingModuleTest.*
is reduced by more than 10x, when run single-threaded.
The PCMFile helper class is extended with a FastForward method (to
skip initial silence in the test files) and a limiter on how much to
read.
BUG=webrtc:2463
R=ivoc@webrtc.org
Review URL: https://codereview.webrtc.org/1513223002 .
Cr-Commit-Position: refs/heads/master@{#10973}
diff --git a/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc b/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
index ef45705..e0222af 100644
--- a/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -63,6 +63,10 @@
if (channels == 2) {
_pcmFile.ReadStereo(true);
}
+ // Set test length to 500 ms (50 blocks of 10 ms each).
+ _pcmFile.SetNum10MsBlocksToRead(50);
+ // Fast-forward 1 second (100 blocks) since the file starts with silence.
+ _pcmFile.FastForward(100);
// Set the codec for the current test.
if ((testMode == 0) || (testMode == 1)) {
diff --git a/webrtc/modules/audio_coding/test/PCMFile.cc b/webrtc/modules/audio_coding/test/PCMFile.cc
index 0466e02..9289d73 100644
--- a/webrtc/modules/audio_coding/test/PCMFile.cc
+++ b/webrtc/modules/audio_coding/test/PCMFile.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "PCMFile.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
#include <ctype.h>
#include <stdio.h>
@@ -137,6 +137,9 @@
audio_frame.num_channels_ = channels;
audio_frame.timestamp_ = timestamp_;
timestamp_ += samples_10ms_;
+ ++blocks_read_;
+ if (num_10ms_blocks_to_read_ && blocks_read_ >= *num_10ms_blocks_to_read_)
+ end_of_file_ = true;
return samples_10ms_;
}
@@ -182,11 +185,21 @@
void PCMFile::Close() {
fclose(pcm_file_);
pcm_file_ = NULL;
+ blocks_read_ = 0;
+}
+
+void PCMFile::FastForward(int num_10ms_blocks) {
+ const int channels = read_stereo_ ? 2 : 1;
+ long num_bytes_to_move =
+ num_10ms_blocks * sizeof(int16_t) * samples_10ms_ * channels;
+ int error = fseek(pcm_file_, num_bytes_to_move, SEEK_CUR);
+ RTC_DCHECK_EQ(error, 0);
}
void PCMFile::Rewind() {
rewind(pcm_file_);
end_of_file_ = false;
+ blocks_read_ = 0;
}
bool PCMFile::Rewinded() {
@@ -201,4 +214,8 @@
read_stereo_ = is_stereo;
}
+void PCMFile::SetNum10MsBlocksToRead(int value) {
+ num_10ms_blocks_to_read_ = rtc::Optional<int>(value);
+}
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/test/PCMFile.h b/webrtc/modules/audio_coding/test/PCMFile.h
index 9365180..840933a 100644
--- a/webrtc/modules/audio_coding/test/PCMFile.h
+++ b/webrtc/modules/audio_coding/test/PCMFile.h
@@ -16,6 +16,7 @@
#include <string>
+#include "webrtc/base/optional.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
@@ -45,12 +46,21 @@
bool EndOfFile() const {
return end_of_file_;
}
+ // Moves forward the specified number of 10 ms blocks. If a limit has been set
+ // with SetNum10MsBlocksToRead, fast-forwarding does not count towards this
+ // limit.
+ void FastForward(int num_10ms_blocks);
void Rewind();
static int16_t ChooseFile(std::string* file_name, int16_t max_len,
uint16_t* frequency_hz);
bool Rewinded();
void SaveStereo(bool is_stereo = true);
void ReadStereo(bool is_stereo = true);
+ // If set, the reading will stop after the specified number of blocks have
+ // been read. When that has happened, EndOfFile() will return true. Calling
+ // Rewind() will reset the counter and start over.
+ void SetNum10MsBlocksToRead(int value);
+
private:
FILE* pcm_file_;
uint16_t samples_10ms_;
@@ -61,6 +71,8 @@
uint32_t timestamp_;
bool read_stereo_;
bool save_stereo_;
+ rtc::Optional<int> num_10ms_blocks_to_read_;
+ int blocks_read_ = 0;
};
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/test/TestAllCodecs.cc b/webrtc/modules/audio_coding/test/TestAllCodecs.cc
index 21ce7c1..bacfd37 100644
--- a/webrtc/modules/audio_coding/test/TestAllCodecs.cc
+++ b/webrtc/modules/audio_coding/test/TestAllCodecs.cc
@@ -422,8 +422,12 @@
uint32_t timestamp_diff;
channel->reset_payload_size();
int error_count = 0;
-
int counter = 0;
+ // Set test length to 500 ms (50 blocks of 10 ms each).
+ infile_a_.SetNum10MsBlocksToRead(50);
+ // Fast-forward 1 second (100 blocks) since the file starts with silence.
+ infile_a_.FastForward(100);
+
while (!infile_a_.EndOfFile()) {
// Add 10 msec to ACM.
infile_a_.Read10MsData(audio_frame);
diff --git a/webrtc/modules/audio_coding/test/TestRedFec.cc b/webrtc/modules/audio_coding/test/TestRedFec.cc
index d544026..a1bdc04 100644
--- a/webrtc/modules/audio_coding/test/TestRedFec.cc
+++ b/webrtc/modules/audio_coding/test/TestRedFec.cc
@@ -453,6 +453,10 @@
void TestRedFec::Run() {
AudioFrame audioFrame;
int32_t outFreqHzB = _outFileB.SamplingFrequency();
+ // Set test length to 500 ms (50 blocks of 10 ms each).
+ _inFileA.SetNum10MsBlocksToRead(50);
+ // Fast-forward 1 second (100 blocks) since the file starts with silence.
+ _inFileA.FastForward(100);
while (!_inFileA.EndOfFile()) {
EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
diff --git a/webrtc/modules/audio_coding/test/TestStereo.cc b/webrtc/modules/audio_coding/test/TestStereo.cc
index 19f027b..9bf560d 100644
--- a/webrtc/modules/audio_coding/test/TestStereo.cc
+++ b/webrtc/modules/audio_coding/test/TestStereo.cc
@@ -735,6 +735,12 @@
int error_count = 0;
int variable_bytes = 0;
int variable_packets = 0;
+ // Set test length to 500 ms (50 blocks of 10 ms each).
+ in_file_mono_->SetNum10MsBlocksToRead(50);
+ in_file_stereo_->SetNum10MsBlocksToRead(50);
+ // Fast-forward 1 second (100 blocks) since the files start with silence.
+ in_file_stereo_->FastForward(100);
+ in_file_mono_->FastForward(100);
while (1) {
// Simulate packet loss by setting |packet_loss_| to "true" in
@@ -800,7 +806,7 @@
// such as Opus.
if (variable_packets > 0) {
variable_bytes /= variable_packets;
- EXPECT_NEAR(variable_bytes, pack_size_bytes_, 3);
+ EXPECT_NEAR(variable_bytes, pack_size_bytes_, 18);
}
if (in_file_mono_->EndOfFile()) {
diff --git a/webrtc/modules/audio_coding/test/TestVADDTX.cc b/webrtc/modules/audio_coding/test/TestVADDTX.cc
index 98b1224..a7d2c21 100644
--- a/webrtc/modules/audio_coding/test/TestVADDTX.cc
+++ b/webrtc/modules/audio_coding/test/TestVADDTX.cc
@@ -87,6 +87,11 @@
PCMFile in_file;
in_file.Open(in_filename, frequency, "rb");
in_file.ReadStereo(channels > 1);
+ // Set test length to 1000 ms (100 blocks of 10 ms each).
+ in_file.SetNum10MsBlocksToRead(100);
+ // Fast-forward both files 500 ms (50 blocks). The first second of the file is
+ // silence, but we want to keep half of that to test silence periods.
+ in_file.FastForward(50);
PCMFile out_file;
if (append) {
diff --git a/webrtc/modules/audio_coding/test/iSACTest.cc b/webrtc/modules/audio_coding/test/iSACTest.cc
index 09744b1..9f223fb 100644
--- a/webrtc/modules/audio_coding/test/iSACTest.cc
+++ b/webrtc/modules/audio_coding/test/iSACTest.cc
@@ -117,6 +117,10 @@
EXPECT_EQ(0, _acmA->RegisterSendCodec(_paramISAC32kHz));
_inFileA.Open(file_name_swb_, 32000, "rb");
+ // Set test length to 500 ms (50 blocks of 10 ms each).
+ _inFileA.SetNum10MsBlocksToRead(50);
+ // Fast-forward 1 second (100 blocks) since the files start with silence.
+ _inFileA.FastForward(100);
std::string fileNameA = webrtc::test::OutputPath() + "testisac_a.pcm";
std::string fileNameB = webrtc::test::OutputPath() + "testisac_b.pcm";
_outFileA.Open(fileNameA, 32000, "wb");
diff --git a/webrtc/modules/audio_coding/test/opus_test.cc b/webrtc/modules/audio_coding/test/opus_test.cc
index 3372a2a..a68db91 100644
--- a/webrtc/modules/audio_coding/test/opus_test.cc
+++ b/webrtc/modules/audio_coding/test/opus_test.cc
@@ -235,8 +235,12 @@
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) {
+ // Fast-forward 1 second (100 blocks) since the files start with silence.
+ in_file_stereo_.FastForward(100);
+ in_file_mono_.FastForward(100);
+
+ // Limit the runtime to 1000 blocks of 10 ms each.
+ for (size_t audio_length = 0; audio_length < 1000; audio_length += 10) {
bool lost_packet = false;
// Get 10 msec of audio.
