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-rw-r--r--modules/BUILD.gn2
-rw-r--r--modules/audio_coding/codecs/isac/audio_encoder_isac_t.h1
-rw-r--r--modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h2
-rw-r--r--modules/audio_coding/codecs/isac/isac_webrtc_api_test.cc82
-rw-r--r--modules/audio_coding/codecs/isac/main/source/structs.h4
-rw-r--r--resources/audio_coding/testfile16kHz.pcm.sha11
6 files changed, 53 insertions, 39 deletions
diff --git a/modules/BUILD.gn b/modules/BUILD.gn
index 2c95fc4f40..ffdd7016a1 100644
--- a/modules/BUILD.gn
+++ b/modules/BUILD.gn
@@ -57,6 +57,7 @@ rtc_source_set("module_fec_api") {
if (rtc_include_tests) {
modules_tests_resources = [
+ "../resources/audio_coding/testfile16kHz.pcm",
"../resources/audio_coding/testfile32kHz.pcm",
"../resources/audio_coding/teststereo32kHz.pcm",
"../resources/foreman_cif.yuv",
@@ -111,6 +112,7 @@ if (rtc_include_tests) {
"../resources/audio_coding/speech_4_channels_48k_one_second.wav",
"../resources/audio_coding/speech_mono_16kHz.pcm",
"../resources/audio_coding/speech_mono_32_48kHz.pcm",
+ "../resources/audio_coding/testfile16kHz.pcm",
"../resources/audio_coding/testfile32kHz.pcm",
"../resources/audio_coding/testfile_fake_stereo_32kHz.pcm",
"../resources/audio_coding/teststereo32kHz.pcm",
diff --git a/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h b/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
index f9a4c97b83..c7b595107f 100644
--- a/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
+++ b/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
@@ -28,7 +28,6 @@ class AudioEncoderIsacT final : public AudioEncoder {
// - 32000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support)
struct Config {
bool IsOk() const;
-
int payload_type = 103;
int sample_rate_hz = 16000;
int frame_size_ms = 30;
diff --git a/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
index 7c0b1a542d..27a02b5006 100644
--- a/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
+++ b/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
@@ -126,7 +126,7 @@ void AudioEncoderIsacT<T>::RecreateEncoderInstance(const Config& config) {
if (isac_state_)
RTC_CHECK_EQ(0, T::Free(isac_state_));
RTC_CHECK_EQ(0, T::Create(&isac_state_));
- RTC_CHECK_EQ(0, T::EncoderInit(isac_state_, 1));
+ RTC_CHECK_EQ(0, T::EncoderInit(isac_state_, /*coding_mode=*/1));
RTC_CHECK_EQ(0, T::SetEncSampRate(isac_state_, config.sample_rate_hz));
const int bit_rate = config.bit_rate == 0 ? kDefaultBitRate : config.bit_rate;
RTC_CHECK_EQ(0, T::Control(isac_state_, bit_rate, config.frame_size_ms));
diff --git a/modules/audio_coding/codecs/isac/isac_webrtc_api_test.cc b/modules/audio_coding/codecs/isac/isac_webrtc_api_test.cc
index 3a2d6f498e..c4d7ab8fa8 100644
--- a/modules/audio_coding/codecs/isac/isac_webrtc_api_test.cc
+++ b/modules/audio_coding/codecs/isac/isac_webrtc_api_test.cc
@@ -9,7 +9,7 @@
*/
#include <array>
-#include <limits>
+#include <memory>
#include <vector>
#include "absl/strings/string_view.h"
@@ -18,9 +18,11 @@
#include "api/audio_codecs/isac/audio_decoder_isac_float.h"
#include "api/audio_codecs/isac/audio_encoder_isac_fix.h"
#include "api/audio_codecs/isac/audio_encoder_isac_float.h"
-#include "rtc_base/random.h"
+#include "modules/audio_coding/test/PCMFile.h"
+#include "rtc_base/checks.h"
#include "rtc_base/strings/string_builder.h"
#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
namespace webrtc {
namespace {
@@ -38,15 +40,31 @@ absl::string_view IsacImplToString(IsacImpl impl) {
}
}
-std::vector<int16_t> GetRandomSamplesVector(size_t size) {
- constexpr int32_t kMin = std::numeric_limits<int16_t>::min();
- constexpr int32_t kMax = std::numeric_limits<int16_t>::max();
- std::vector<int16_t> v(size);
- Random gen(/*seed=*/42);
- for (auto& x : v) {
- x = static_cast<int16_t>(gen.Rand(kMin, kMax));
+std::unique_ptr<PCMFile> GetPcmTestFileReader(int sample_rate_hz) {
+ std::string filename;
+ switch (sample_rate_hz) {
+ case 16000:
+ filename = test::ResourcePath("audio_coding/testfile16kHz", "pcm");
+ break;
+ case 32000:
+ filename = test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+ break;
+ default:
+ RTC_NOTREACHED() << "No test file available for " << sample_rate_hz
+ << " Hz.";
}
- return v;
+ auto pcm_file = std::make_unique<PCMFile>();
+ pcm_file->ReadStereo(false);
+ pcm_file->Open(filename, sample_rate_hz, "rb", /*auto_rewind=*/true);
+ pcm_file->FastForward(/*num_10ms_blocks=*/100); // Skip initial silence.
+ RTC_CHECK(!pcm_file->EndOfFile());
+ return pcm_file;
+}
+
+// Returns a view to the interleaved samples of an AudioFrame object.
+rtc::ArrayView<const int16_t> AudioFrameToView(const AudioFrame& audio_frame) {
+ return {audio_frame.data(),
+ audio_frame.samples_per_channel() * audio_frame.num_channels()};
}
std::unique_ptr<AudioEncoder> CreateEncoder(IsacImpl impl,
@@ -119,6 +137,7 @@ TEST_P(EncoderTest, TestConfig) {
// checks that the number of produces bytes in the first case is less than that
// of the second case.
TEST_P(EncoderTest, TestDifferentBitrates) {
+ auto pcm_file = GetPcmTestFileReader(GetSampleRateHz());
constexpr int kLowBps = 20000;
constexpr int kHighBps = 25000;
auto encoder_low = CreateEncoder(GetIsacImpl(), GetSampleRateHz(),
@@ -127,39 +146,39 @@ TEST_P(EncoderTest, TestDifferentBitrates) {
GetFrameSizeMs(), kHighBps);
int num_bytes_low = 0;
int num_bytes_high = 0;
- const auto in = GetRandomSamplesVector(
- /*size=*/rtc::CheckedDivExact(GetSampleRateHz(), 100));
constexpr int kNumFrames = 12;
for (int i = 0; i < kNumFrames; ++i) {
+ AudioFrame in;
+ pcm_file->Read10MsData(in);
rtc::Buffer low, high;
- encoder_low->Encode(/*rtp_timestamp=*/0, in, &low);
- encoder_high->Encode(/*rtp_timestamp=*/0, in, &high);
+ encoder_low->Encode(/*rtp_timestamp=*/0, AudioFrameToView(in), &low);
+ encoder_high->Encode(/*rtp_timestamp=*/0, AudioFrameToView(in), &high);
num_bytes_low += low.size();
num_bytes_high += high.size();
}
EXPECT_LT(num_bytes_low, num_bytes_high);
}
-// Checks that the target and the measured bitrates are within tolerance.
-// TODO(webrtc:11360): Add CBR flag to the config and re-enable test with CBR.
-TEST_P(EncoderTest, DISABLED_TestBitrateNearTarget) {
- const auto in = GetRandomSamplesVector(
- /*size=*/rtc::CheckedDivExact(GetSampleRateHz(), 100)); // 10 ms.
+// Checks that, given a target bitrate, the encoder does not overshoot too much.
+TEST_P(EncoderTest, DoNotOvershootTargetBitrate) {
for (int bitrate_bps : {10000, 15000, 20000, 26000, 32000}) {
SCOPED_TRACE(bitrate_bps);
+ auto pcm_file = GetPcmTestFileReader(GetSampleRateHz());
auto e = CreateEncoder(GetIsacImpl(), GetSampleRateHz(), GetFrameSizeMs(),
bitrate_bps);
int num_bytes = 0;
- constexpr int kNumFrames = 60;
+ constexpr int kNumFrames = 200; // 2 seconds.
for (int i = 0; i < kNumFrames; ++i) {
+ AudioFrame in;
+ pcm_file->Read10MsData(in);
rtc::Buffer encoded;
- e->Encode(/*rtp_timestamp=*/0, in, &encoded);
+ e->Encode(/*rtp_timestamp=*/0, AudioFrameToView(in), &encoded);
num_bytes += encoded.size();
}
// Inverse of the duration of |kNumFrames| 10 ms frames (unit: seconds^-1).
constexpr float kAudioDurationInv = 100.f / kNumFrames;
const int measured_bitrate_bps = 8 * num_bytes * kAudioDurationInv;
- EXPECT_NEAR(bitrate_bps, measured_bitrate_bps, 1000); // Max 1 kbps.
+ EXPECT_LT(measured_bitrate_bps, bitrate_bps + 2000); // Max 2 kbps extra.
}
}
@@ -227,27 +246,19 @@ struct EncoderDecoderPairTestParams {
class EncoderDecoderPairTest
: public testing::TestWithParam<EncoderDecoderPairTestParams> {
protected:
- EncoderDecoderPairTest()
- : input_frame_(GetRandomSamplesVector(GetInputFrameSize())) {}
- rtc::ArrayView<const int16_t> GetInputFrame() { return input_frame_; }
+ EncoderDecoderPairTest() = default;
int GetSampleRateHz() const { return GetParam().sample_rate_hz; }
int GetEncoderFrameSizeMs() const { return GetParam().frame_size_ms; }
IsacImpl GetEncoderIsacImpl() const { return GetParam().encoder_impl; }
IsacImpl GetDecoderIsacImpl() const { return GetParam().decoder_impl; }
-
int GetEncoderFrameSize() const {
return GetEncoderFrameSizeMs() * GetSampleRateHz() / 1000;
}
-
- private:
- const std::vector<int16_t> input_frame_;
- int GetInputFrameSize() const {
- return rtc::CheckedDivExact(GetParam().sample_rate_hz, 100); // 10 ms.
- }
};
// Checks that the number of encoded and decoded samples match.
TEST_P(EncoderDecoderPairTest, EncodeDecode) {
+ auto pcm_file = GetPcmTestFileReader(GetSampleRateHz());
auto encoder = CreateEncoder(GetEncoderIsacImpl(), GetSampleRateHz(),
GetEncoderFrameSizeMs(), /*bitrate_bps=*/20000);
auto decoder = CreateDecoder(GetDecoderIsacImpl(), GetSampleRateHz());
@@ -257,10 +268,11 @@ TEST_P(EncoderDecoderPairTest, EncodeDecode) {
size_t num_decoded_samples = 0;
constexpr int kNumFrames = 12;
for (int i = 0; i < kNumFrames; ++i) {
+ AudioFrame in;
+ pcm_file->Read10MsData(in);
rtc::Buffer encoded;
- auto in = GetInputFrame();
- encoder->Encode(/*rtp_timestamp=*/0, in, &encoded);
- num_encoded_samples += in.size();
+ encoder->Encode(/*rtp_timestamp=*/0, AudioFrameToView(in), &encoded);
+ num_encoded_samples += in.samples_per_channel();
if (encoded.empty()) {
continue;
}
diff --git a/modules/audio_coding/codecs/isac/main/source/structs.h b/modules/audio_coding/codecs/isac/main/source/structs.h
index 1255d5b16c..6861ca42bd 100644
--- a/modules/audio_coding/codecs/isac/main/source/structs.h
+++ b/modules/audio_coding/codecs/isac/main/source/structs.h
@@ -304,7 +304,7 @@ typedef struct {
int16_t maxRateInBytes;
/*---
- If set to 1 iSAC will not addapt the frame-size, if used in
+ If set to 1 iSAC will not adapt the frame-size, if used in
channel-adaptive mode. The initial value will be used for all rates.
---*/
int16_t enforceFrameSize;
@@ -312,7 +312,7 @@ typedef struct {
/*-----
This records the BWE index the encoder injected into the bit-stream.
It will be used in RCU. The same BWE index of main payload will be in
- the redundant payload. We can not retrive it from BWE because it is
+ the redundant payload. We can not retrieve it from BWE because it is
a recursive procedure (WebRtcIsac_GetDownlinkBwJitIndexImpl) and has to be
called only once per each encode.
-----*/
diff --git a/resources/audio_coding/testfile16kHz.pcm.sha1 b/resources/audio_coding/testfile16kHz.pcm.sha1
new file mode 100644
index 0000000000..4d7df3be8f
--- /dev/null
+++ b/resources/audio_coding/testfile16kHz.pcm.sha1
@@ -0,0 +1 @@
+0d2702e5c350c2a4ad3a641c4d96271e8aa12e6c \ No newline at end of file
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-01 16:00:08 +0000