iSAC unit test: test encode/decode via API wrapper
Unit test to test the iSAC webrtc API wrapper, plus a minor
change in the c iSAC wrapper.
Bug: webrtc:10584
Change-Id: Iecbf6f3e7db5b3bdba41f8428254ae6a6a73e24a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168492
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30514}
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 669deeb..8efc221 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -744,7 +744,8 @@
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
- public_deps = [ ":webrtc_opus_wrapper" ] # no-presubmit-check TODO(webrtc:8603)
+ public_deps = # no-presubmit-check TODO(webrtc:8603)
+ [ ":webrtc_opus_wrapper" ]
defines = audio_codec_defines
@@ -780,7 +781,8 @@
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
- public_deps = [ ":webrtc_opus_wrapper" ] # no-presubmit-check TODO(webrtc:8603)
+ public_deps = # no-presubmit-check TODO(webrtc:8603)
+ [ ":webrtc_opus_wrapper" ]
defines = audio_codec_defines
@@ -865,7 +867,8 @@
"audio_network_adaptor/util/threshold_curve.h",
]
- public_deps = [ ":audio_network_adaptor_config" ] # no-presubmit-check TODO(webrtc:8603)
+ public_deps = # no-presubmit-check TODO(webrtc:8603)
+ [ ":audio_network_adaptor_config" ]
deps = [
"../../api/audio_codecs:audio_codecs_api",
@@ -1160,7 +1163,8 @@
"../rtp_rtcp:rtp_rtcp_format",
"//third_party/abseil-cpp/absl/types:optional",
]
- public_deps = [ "../../logging:rtc_event_log_proto" ] # no-presubmit-check TODO(webrtc:8603)
+ public_deps = # no-presubmit-check TODO(webrtc:8603)
+ [ "../../logging:rtc_event_log_proto" ]
}
# Only used for test purpose. Since we want to use it from chromium
@@ -1911,6 +1915,7 @@
"codecs/isac/fix/source/filters_unittest.cc",
"codecs/isac/fix/source/lpc_masking_model_unittest.cc",
"codecs/isac/fix/source/transform_unittest.cc",
+ "codecs/isac/isac_webrtc_api_test.cc",
"codecs/isac/main/source/audio_encoder_isac_unittest.cc",
"codecs/isac/main/source/isac_unittest.cc",
"codecs/legacy_encoded_audio_frame_unittest.cc",
@@ -1976,6 +1981,7 @@
":ilbc",
":isac",
":isac_c",
+ ":isac_common",
":isac_fix",
":legacy_encoded_audio_frame",
":mocks",
@@ -1988,10 +1994,15 @@
":webrtc_opus",
"..:module_api",
"..:module_api_public",
+ "../../api:array_view",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../api/audio_codecs:builtin_audio_encoder_factory",
+ "../../api/audio_codecs/isac:audio_decoder_isac_fix",
+ "../../api/audio_codecs/isac:audio_decoder_isac_float",
+ "../../api/audio_codecs/isac:audio_encoder_isac_fix",
+ "../../api/audio_codecs/isac:audio_encoder_isac_float",
"../../api/audio_codecs/opus:audio_decoder_multiopus",
"../../api/audio_codecs/opus:audio_decoder_opus",
"../../api/audio_codecs/opus:audio_encoder_multiopus",
diff --git a/modules/audio_coding/codecs/isac/isac_webrtc_api_test.cc b/modules/audio_coding/codecs/isac/isac_webrtc_api_test.cc
new file mode 100644
index 0000000..ac83861
--- /dev/null
+++ b/modules/audio_coding/codecs/isac/isac_webrtc_api_test.cc
@@ -0,0 +1,145 @@
+/*
+ * Copyright (c) 2020 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 <array>
+#include <limits>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/audio_codecs/isac/audio_decoder_isac_fix.h"
+#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 "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kPayloadType = 42;
+constexpr int kBitrateBps = 20000;
+
+enum class IsacImpl { kFixed, kFloat };
+
+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));
+ }
+ return v;
+}
+
+class IsacApiTest
+ : public testing::TestWithParam<std::tuple<int, int, IsacImpl, IsacImpl>> {
+ protected:
+ IsacApiTest() : input_frame_(GetRandomSamplesVector(GetInputFrameLength())) {}
+ rtc::ArrayView<const int16_t> GetInputFrame() { return input_frame_; }
+ int GetSampleRateHz() const { return std::get<0>(GetParam()); }
+ int GetEncoderFrameLenght() const {
+ return GetEncoderFrameLenghtMs() * GetSampleRateHz() / 1000;
+ }
+ std::unique_ptr<AudioEncoder> CreateEncoder() const {
+ switch (GetEncoderIsacImpl()) {
+ case IsacImpl::kFixed: {
+ AudioEncoderIsacFix::Config config;
+ config.frame_size_ms = GetEncoderFrameLenghtMs();
+ RTC_CHECK_EQ(16000, GetSampleRateHz());
+ return AudioEncoderIsacFix::MakeAudioEncoder(config, kPayloadType);
+ }
+ case IsacImpl::kFloat: {
+ AudioEncoderIsacFloat::Config config;
+ config.bit_rate = kBitrateBps;
+ config.frame_size_ms = GetEncoderFrameLenghtMs();
+ config.sample_rate_hz = GetSampleRateHz();
+ return AudioEncoderIsacFloat::MakeAudioEncoder(config, kPayloadType);
+ }
+ }
+ }
+ std::unique_ptr<AudioDecoder> CreateDecoder() const {
+ switch (GetDecoderIsacImpl()) {
+ case IsacImpl::kFixed: {
+ webrtc::AudioDecoderIsacFix::Config config;
+ RTC_CHECK_EQ(16000, GetSampleRateHz());
+ return webrtc::AudioDecoderIsacFix::MakeAudioDecoder(config);
+ }
+ case IsacImpl::kFloat: {
+ webrtc::AudioDecoderIsacFloat::Config config;
+ config.sample_rate_hz = GetSampleRateHz();
+ return webrtc::AudioDecoderIsacFloat::MakeAudioDecoder(config);
+ }
+ }
+ }
+
+ private:
+ const std::vector<int16_t> input_frame_;
+ int GetInputFrameLength() const {
+ return rtc::CheckedDivExact(std::get<0>(GetParam()), 100); // 10 ms.
+ }
+ int GetEncoderFrameLenghtMs() const {
+ int frame_size_ms = std::get<1>(GetParam());
+ RTC_CHECK(frame_size_ms == 30 || frame_size_ms == 60);
+ return frame_size_ms;
+ }
+ IsacImpl GetEncoderIsacImpl() const { return std::get<2>(GetParam()); }
+ IsacImpl GetDecoderIsacImpl() const { return std::get<3>(GetParam()); }
+};
+
+// Checks that the number of encoded and decoded samples match.
+TEST_P(IsacApiTest, EncodeDecode) {
+ auto encoder = CreateEncoder();
+ auto decoder = CreateDecoder();
+ const int encoder_frame_length = GetEncoderFrameLenght();
+ std::vector<int16_t> out(encoder_frame_length);
+ size_t num_encoded_samples = 0;
+ size_t num_decoded_samples = 0;
+ constexpr int kNumFrames = 12;
+ for (int i = 0; i < kNumFrames; ++i) {
+ rtc::Buffer encoded;
+ auto in = GetInputFrame();
+ encoder->Encode(/*rtp_timestamp=*/0, in, &encoded);
+ num_encoded_samples += in.size();
+ if (encoded.empty()) {
+ continue;
+ }
+ // Decode.
+ const std::vector<AudioDecoder::ParseResult> parse_result =
+ decoder->ParsePayload(std::move(encoded), /*timestamp=*/0);
+ EXPECT_EQ(parse_result.size(), size_t{1});
+ auto decode_result = parse_result[0].frame->Decode(out);
+ EXPECT_TRUE(decode_result.has_value());
+ EXPECT_EQ(out.size(), decode_result->num_decoded_samples);
+ num_decoded_samples += decode_result->num_decoded_samples;
+ }
+ EXPECT_EQ(num_encoded_samples, num_decoded_samples);
+}
+
+// Creates tests for different encoder frame lengths and different
+// encoder/decoder implementations.
+INSTANTIATE_TEST_SUITE_P(
+ AllTest,
+ IsacApiTest,
+ ::testing::ValuesIn([] {
+ std::vector<std::tuple<int, int, IsacImpl, IsacImpl>> cases;
+ for (int frame_length_ms : {30, 60}) {
+ for (IsacImpl enc : {IsacImpl::kFloat, IsacImpl::kFixed}) {
+ for (IsacImpl dec : {IsacImpl::kFloat, IsacImpl::kFixed}) {
+ cases.push_back({16000, frame_length_ms, enc, dec});
+ }
+ }
+ }
+ cases.push_back({32000, 30, IsacImpl::kFloat, IsacImpl::kFloat});
+ return cases;
+ }()));
+
+} // namespace
+} // namespace webrtc
diff --git a/modules/audio_coding/codecs/isac/main/include/isac.h b/modules/audio_coding/codecs/isac/main/include/isac.h
index 6bbbf8a..3d2caef 100644
--- a/modules/audio_coding/codecs/isac/main/include/isac.h
+++ b/modules/audio_coding/codecs/isac/main/include/isac.h
@@ -252,7 +252,7 @@
*
*/
-int16_t WebRtcIsac_ReadFrameLen(ISACStruct* ISAC_main_inst,
+int16_t WebRtcIsac_ReadFrameLen(const ISACStruct* ISAC_main_inst,
const uint8_t* encoded,
int16_t* frameLength);
diff --git a/modules/audio_coding/codecs/isac/main/source/isac.c b/modules/audio_coding/codecs/isac/main/source/isac.c
index 552bab8..73f132c 100644
--- a/modules/audio_coding/codecs/isac/main/source/isac.c
+++ b/modules/audio_coding/codecs/isac/main/source/isac.c
@@ -1719,7 +1719,7 @@
* - frameLength : Length of frame in packet (in samples)
*
*/
-int16_t WebRtcIsac_ReadFrameLen(ISACStruct* ISAC_main_inst,
+int16_t WebRtcIsac_ReadFrameLen(const ISACStruct* ISAC_main_inst,
const uint8_t* encoded,
int16_t* frameLength) {
Bitstr streamdata;