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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-11 08:17:27 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-11 08:17:27 +0000
commitf215e02bf85f68d3a6106c2a1f4f7f063f819064 (patch)
tree6bb5b92c046312c4e95ac2620b10ddf482d3fa8b /src/VBox/Devices/Audio/testcase
parentInitial commit. (diff)
downloadvirtualbox-f215e02bf85f68d3a6106c2a1f4f7f063f819064.tar.xz
virtualbox-f215e02bf85f68d3a6106c2a1f4f7f063f819064.zip
Adding upstream version 7.0.14-dfsg.upstream/7.0.14-dfsg
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'src/VBox/Devices/Audio/testcase')
-rw-r--r--src/VBox/Devices/Audio/testcase/Makefile.kmk83
-rw-r--r--src/VBox/Devices/Audio/testcase/tstAudioClient3.cpp112
-rw-r--r--src/VBox/Devices/Audio/testcase/tstAudioMixBuffer.cpp916
-rw-r--r--src/VBox/Devices/Audio/testcase/tstAudioTestService.cpp178
4 files changed, 1289 insertions, 0 deletions
diff --git a/src/VBox/Devices/Audio/testcase/Makefile.kmk b/src/VBox/Devices/Audio/testcase/Makefile.kmk
new file mode 100644
index 00000000..39cd4109
--- /dev/null
+++ b/src/VBox/Devices/Audio/testcase/Makefile.kmk
@@ -0,0 +1,83 @@
+# $Id: Makefile.kmk $
+## @file
+# Sub-Makefile for the audio testcases.
+#
+
+#
+# Copyright (C) 2014-2023 Oracle and/or its affiliates.
+#
+# This file is part of VirtualBox base platform packages, as
+# available from https://www.virtualbox.org.
+#
+# This program is free software; you can redistribute it and/or
+# modify it under the terms of the GNU General Public License
+# as published by the Free Software Foundation, in version 3 of the
+# License.
+#
+# This program is distributed in the hope that it will be useful, but
+# WITHOUT ANY WARRANTY; without even the implied warranty of
+# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+# General Public License for more details.
+#
+# You should have received a copy of the GNU General Public License
+# along with this program; if not, see <https://www.gnu.org/licenses>.
+#
+# SPDX-License-Identifier: GPL-3.0-only
+#
+
+SUB_DEPTH = ../../../../..
+include $(KBUILD_PATH)/subheader.kmk
+
+if defined(VBOX_WITH_TESTCASES) && !defined(VBOX_ONLY_ADDITIONS) && !defined(VBOX_ONLY_SDK)
+
+ PROGRAMS += tstAudioMixBuffer
+ TESTING += $(tstAudioMixBuffer_0_OUTDIR)/tstAudioMixBuffer.run
+
+ tstAudioMixBuffer_TEMPLATE = VBoxR3TstExe
+ tstAudioMixBuffer_DEFS = TESTCASE VBOX_AUDIO_MIX_BUFFER_TESTCASE
+ tstAudioMixBuffer_DEFS.debug = VBOX_WITH_EF_WRAPS
+ tstAudioMixBuffer_SOURCES = \
+ tstAudioMixBuffer.cpp \
+ ../AudioMixBuffer.cpp \
+ ../AudioHlp.cpp
+ tstAudioMixBuffer_LIBS = $(LIB_RUNTIME)
+
+ $$(tstAudioMixBuffer_0_OUTDIR)/tstAudioMixBuffer.run: $$(tstAudioMixBuffer_1_STAGE_TARGET)
+ export VBOX_LOG_DEST=nofile; $(tstAudioMixBuffer_1_STAGE_TARGET) quiet
+ $(QUIET)$(APPEND) -t "$@" "done"
+
+ ifeq ($(KBUILD_TARGET),win.amd64) # Note: Only runs on Windows 8 or newer.
+ tstAudioClient3_TEMPLATE = VBoxR3TstExe
+ tstAudioClient3_DEFS = TESTCASE
+ tstAudioClient3_DEFS.debug = VBOX_WITH_EF_WRAPS
+ tstAudioClient3_SOURCES = tstAudioClient3.cpp
+ tstAudioClient3_LIBS = $(LIB_RUNTIME)
+ tstAudioClient3_INCS = \
+ $(KBUILD_DEVTOOLS)/win.x86/sdk/v10.0.17134.0/Include/10.0.17134.0/um \
+ $(KBUILD_DEVTOOLS)/win.x86/sdk/v10.0.17134.0/include/10.0.17134.0/shared
+
+ # Requires manual execution / verification.
+ PROGRAMS += tstAudioClient3
+ endif
+
+ PROGRAMS += tstAudioTestService
+ TESTING += $(tstAudioTestService_0_OUTDIR)/tstAudioTestService.run
+
+ tstAudioTestService_TEMPLATE = VBoxR3TstExe
+ tstAudioTestService_DEFS = TESTCASE
+ tstAudioTestService_DEFS.debug = VBOX_WITH_EF_WRAPS
+ tstAudioTestService_SOURCES = \
+ tstAudioTestService.cpp \
+ ../AudioTestService.cpp \
+ ../AudioTestServiceProtocol.cpp \
+ ../AudioTestServiceTcp.cpp \
+ ../AudioTestServiceClient.cpp
+ tstAudioTestService_LIBS = $(LIB_RUNTIME)
+
+ $$(tstAudioTestService_0_OUTDIR)/tstAudioTestService.run: $$(tstAudioTestService_1_STAGE_TARGET)
+ export VBOX_LOG_DEST=nofile; $(tstAudioTestService_1_STAGE_TARGET) quiet
+ $(QUIET)$(APPEND) -t "$@" "done"
+
+endif
+
+include $(FILE_KBUILD_SUB_FOOTER)
diff --git a/src/VBox/Devices/Audio/testcase/tstAudioClient3.cpp b/src/VBox/Devices/Audio/testcase/tstAudioClient3.cpp
new file mode 100644
index 00000000..c259e42b
--- /dev/null
+++ b/src/VBox/Devices/Audio/testcase/tstAudioClient3.cpp
@@ -0,0 +1,112 @@
+/* $Id: tstAudioClient3.cpp $ */
+/** @file
+ * Audio testcase - Tests for the IAudioClient3 interface.
+ */
+
+/*
+ * Copyright (C) 2021-2023 Oracle and/or its affiliates.
+ *
+ * This file is part of VirtualBox base platform packages, as
+ * available from https://www.virtualbox.org.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation, in version 3 of the
+ * License.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, see <https://www.gnu.org/licenses>.
+ *
+ * SPDX-License-Identifier: GPL-3.0-only
+ */
+
+
+/*********************************************************************************************************************************
+* Header Files *
+*********************************************************************************************************************************/
+
+#include <iprt/errcore.h>
+#include <iprt/initterm.h>
+#include <iprt/mem.h>
+#include <iprt/rand.h>
+#include <iprt/stream.h>
+#include <iprt/string.h>
+#include <iprt/test.h>
+
+#include <iprt/win/windows.h>
+
+#include <Audioclient.h>
+#include <mmdeviceapi.h>
+
+int main(int argc, char **argv)
+{
+ RTR3InitExe(argc, &argv, 0);
+
+ /*
+ * Initialize IPRT and create the test.
+ */
+ RTTEST hTest;
+ int rc = RTTestInitAndCreate("tstAudioMixBuffer", &hTest);
+ if (rc)
+ return rc;
+ RTTestBanner(hTest);
+
+ /* Note: IAudioClient3 is supported on Win8 or newer. */
+
+ /** @todo Very crude for now, lacks error checking and such. Later. */
+
+ HRESULT hr = CoInitialize(NULL);
+
+ IMMDeviceEnumerator* pEnumerator;
+ hr = CoCreateInstance(__uuidof(MMDeviceEnumerator), NULL, CLSCTX_ALL,
+ __uuidof(IMMDeviceEnumerator),
+ reinterpret_cast<void**>(&pEnumerator));
+
+ IMMDevice* pDevice;
+ hr = pEnumerator->GetDefaultAudioEndpoint(eRender, eConsole, &pDevice);
+
+ IAudioClient3* pAudioClient;
+ hr = pDevice->Activate(__uuidof(IAudioClient3), CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pAudioClient));
+
+ WAVEFORMATEX* pFormat;
+ hr = pAudioClient->GetMixFormat(&pFormat);
+
+ UINT32 defaultPeriodInFrames;
+ UINT32 fundamentalPeriodInFrames;
+ UINT32 minPeriodInFrames;
+ UINT32 maxPeriodInFrames;
+ hr = pAudioClient->GetSharedModeEnginePeriod(pFormat,
+ &defaultPeriodInFrames,
+ &fundamentalPeriodInFrames,
+ &minPeriodInFrames,
+ &maxPeriodInFrames);
+
+ RTTestPrintf(hTest, RTTESTLVL_ALWAYS, "def=%RU32, fundamental=%RU32, min=%RU32, max=%RU32\n",
+ defaultPeriodInFrames, fundamentalPeriodInFrames, minPeriodInFrames, maxPeriodInFrames);
+
+ uint32_t cfDefault = defaultPeriodInFrames * 2;
+
+ RTTestPrintf(hTest, RTTESTLVL_ALWAYS, "Trying to set %RU32 as default ...\n", cfDefault);
+
+ hr = pAudioClient->InitializeSharedAudioStream(0, cfDefault, pFormat, NULL);
+ if (hr != S_OK)
+ RTTestPrintf(hTest, RTTESTLVL_ALWAYS, "Unable to set new period");
+ else
+ {
+ RTTestPrintf(hTest, RTTESTLVL_ALWAYS, "OK");
+
+ hr = pAudioClient->Start();
+
+ /** @todo Do some waiting / testing here. */
+ }
+
+ /*
+ * Summary
+ */
+ return RTTestSummaryAndDestroy(hTest);
+}
diff --git a/src/VBox/Devices/Audio/testcase/tstAudioMixBuffer.cpp b/src/VBox/Devices/Audio/testcase/tstAudioMixBuffer.cpp
new file mode 100644
index 00000000..38f4a9d7
--- /dev/null
+++ b/src/VBox/Devices/Audio/testcase/tstAudioMixBuffer.cpp
@@ -0,0 +1,916 @@
+/* $Id: tstAudioMixBuffer.cpp $ */
+/** @file
+ * Audio testcase - Mixing buffer.
+ */
+
+/*
+ * Copyright (C) 2014-2023 Oracle and/or its affiliates.
+ *
+ * This file is part of VirtualBox base platform packages, as
+ * available from https://www.virtualbox.org.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation, in version 3 of the
+ * License.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, see <https://www.gnu.org/licenses>.
+ *
+ * SPDX-License-Identifier: GPL-3.0-only
+ */
+
+
+/*********************************************************************************************************************************
+* Header Files *
+*********************************************************************************************************************************/
+#include <iprt/errcore.h>
+#include <iprt/initterm.h>
+#include <iprt/mem.h>
+#include <iprt/rand.h>
+#include <iprt/stream.h>
+#include <iprt/string.h>
+#include <iprt/test.h>
+
+#include <VBox/vmm/pdm.h>
+#include <VBox/vmm/pdmaudioinline.h>
+
+#include "../AudioMixBuffer.h"
+#include "../AudioHlp.h"
+
+#define _USE_MATH_DEFINES
+#include <math.h> /* sin, M_PI */
+
+
+/*********************************************************************************************************************************
+* Global Variables *
+*********************************************************************************************************************************/
+#ifdef RT_LITTLE_ENDIAN
+bool const g_fLittleEndian = true;
+#else
+bool const g_fLittleEndian = false;
+#endif
+
+
+static void tstBasics(RTTEST hTest)
+{
+ RTTestSub(hTest, "Basics");
+
+ const PDMAUDIOPCMPROPS Cfg441StereoS16 = PDMAUDIOPCMPROPS_INITIALIZER(
+ /* a_cb: */ 2,
+ /* a_fSigned: */ true,
+ /* a_cChannels: */ 2,
+ /* a_uHz: */ 44100,
+ /* a_fSwapEndian: */ false
+ );
+ const PDMAUDIOPCMPROPS Cfg441StereoU16 = PDMAUDIOPCMPROPS_INITIALIZER(
+ /* a_cb: */ 2,
+ /* a_fSigned: */ false,
+ /* a_cChannels: */ 2,
+ /* a_uHz: */ 44100,
+ /* a_fSwapEndian: */ false
+ );
+ const PDMAUDIOPCMPROPS Cfg441StereoU32 = PDMAUDIOPCMPROPS_INITIALIZER(
+ /* a_cb: */ 4,
+ /* a_fSigned: */ false,
+ /* a_cChannels: */ 2,
+ /* a_uHz: */ 44100,
+ /* a_fSwapEndian: */ false
+ );
+
+ RTTESTI_CHECK(PDMAudioPropsGetBitrate(&Cfg441StereoS16) == 44100*4*8);
+ RTTESTI_CHECK(PDMAudioPropsGetBitrate(&Cfg441StereoU16) == 44100*4*8);
+ RTTESTI_CHECK(PDMAudioPropsGetBitrate(&Cfg441StereoU32) == 44100*8*8);
+
+ RTTESTI_CHECK(AudioHlpPcmPropsAreValidAndSupported(&Cfg441StereoS16));
+ RTTESTI_CHECK(AudioHlpPcmPropsAreValidAndSupported(&Cfg441StereoU16));
+ RTTESTI_CHECK(AudioHlpPcmPropsAreValidAndSupported(&Cfg441StereoU32));
+
+
+ RTTESTI_CHECK_MSG(PDMAUDIOPCMPROPS_F2B(&Cfg441StereoS16, 1) == 4,
+ ("got %x, expected 4\n", PDMAUDIOPCMPROPS_F2B(&Cfg441StereoS16, 1)));
+ RTTESTI_CHECK_MSG(PDMAUDIOPCMPROPS_F2B(&Cfg441StereoU16, 1) == 4,
+ ("got %x, expected 4\n", PDMAUDIOPCMPROPS_F2B(&Cfg441StereoU16, 1)));
+ RTTESTI_CHECK_MSG(PDMAUDIOPCMPROPS_F2B(&Cfg441StereoU32, 1) == 8,
+ ("got %x, expected 4\n", PDMAUDIOPCMPROPS_F2B(&Cfg441StereoU32, 1)));
+
+ RTTESTI_CHECK_MSG(PDMAudioPropsBytesPerFrame(&Cfg441StereoS16) == 4,
+ ("got %x, expected 4\n", PDMAudioPropsBytesPerFrame(&Cfg441StereoS16)));
+ RTTESTI_CHECK_MSG(PDMAudioPropsBytesPerFrame(&Cfg441StereoU16) == 4,
+ ("got %x, expected 4\n", PDMAudioPropsBytesPerFrame(&Cfg441StereoU16)));
+ RTTESTI_CHECK_MSG(PDMAudioPropsBytesPerFrame(&Cfg441StereoU32) == 8,
+ ("got %x, expected 4\n", PDMAudioPropsBytesPerFrame(&Cfg441StereoU32)));
+
+ uint32_t u32;
+ for (uint32_t i = 0; i < 256; i += 8)
+ {
+ RTTESTI_CHECK(PDMAudioPropsIsSizeAligned(&Cfg441StereoU32, i) == true);
+ for (uint32_t j = 1; j < 8; j++)
+ RTTESTI_CHECK(PDMAudioPropsIsSizeAligned(&Cfg441StereoU32, i + j) == false);
+ for (uint32_t j = 0; j < 8; j++)
+ RTTESTI_CHECK(PDMAudioPropsFloorBytesToFrame(&Cfg441StereoU32, i + j) == i);
+ }
+ for (uint32_t i = 0; i < 4096; i += 4)
+ {
+ RTTESTI_CHECK(PDMAudioPropsIsSizeAligned(&Cfg441StereoS16, i) == true);
+ for (uint32_t j = 1; j < 4; j++)
+ RTTESTI_CHECK(PDMAudioPropsIsSizeAligned(&Cfg441StereoS16, i + j) == false);
+ for (uint32_t j = 0; j < 4; j++)
+ RTTESTI_CHECK(PDMAudioPropsFloorBytesToFrame(&Cfg441StereoS16, i + j) == i);
+ }
+
+ RTTESTI_CHECK_MSG((u32 = PDMAudioPropsFramesToBytes(&Cfg441StereoS16, 44100)) == 44100 * 2 * 2,
+ ("cb=%RU32\n", u32));
+ RTTESTI_CHECK_MSG((u32 = PDMAudioPropsFramesToBytes(&Cfg441StereoS16, 2)) == 2 * 2 * 2,
+ ("cb=%RU32\n", u32));
+ RTTESTI_CHECK_MSG((u32 = PDMAudioPropsFramesToBytes(&Cfg441StereoS16, 1)) == 4,
+ ("cb=%RU32\n", u32));
+ RTTESTI_CHECK_MSG((u32 = PDMAudioPropsFramesToBytes(&Cfg441StereoU16, 1)) == 4,
+ ("cb=%RU32\n", u32));
+ RTTESTI_CHECK_MSG((u32 = PDMAudioPropsFramesToBytes(&Cfg441StereoU32, 1)) == 8,
+ ("cb=%RU32\n", u32));
+
+ RTTESTI_CHECK_MSG((u32 = PDMAudioPropsBytesToFrames(&Cfg441StereoS16, 4)) == 1, ("cb=%RU32\n", u32));
+ RTTESTI_CHECK_MSG((u32 = PDMAudioPropsBytesToFrames(&Cfg441StereoU16, 4)) == 1, ("cb=%RU32\n", u32));
+ RTTESTI_CHECK_MSG((u32 = PDMAudioPropsBytesToFrames(&Cfg441StereoU32, 8)) == 1, ("cb=%RU32\n", u32));
+
+ uint64_t u64;
+ RTTESTI_CHECK_MSG((u64 = PDMAudioPropsBytesToNano(&Cfg441StereoS16, 44100 * 2 * 2)) == RT_NS_1SEC,
+ ("ns=%RU64\n", u64));
+ RTTESTI_CHECK_MSG((u64 = PDMAudioPropsBytesToMicro(&Cfg441StereoS16, 44100 * 2 * 2)) == RT_US_1SEC,
+ ("us=%RU64\n", u64));
+ RTTESTI_CHECK_MSG((u64 = PDMAudioPropsBytesToMilli(&Cfg441StereoS16, 44100 * 2 * 2)) == RT_MS_1SEC,
+ ("ms=%RU64\n", u64));
+
+ RTTESTI_CHECK_MSG((u64 = PDMAudioPropsFramesToNano(&Cfg441StereoS16, 44100)) == RT_NS_1SEC, ("ns=%RU64\n", u64));
+ RTTESTI_CHECK_MSG((u64 = PDMAudioPropsFramesToNano(&Cfg441StereoS16, 1)) == 22675, ("ns=%RU64\n", u64));
+ RTTESTI_CHECK_MSG((u64 = PDMAudioPropsFramesToNano(&Cfg441StereoS16, 31)) == 702947, ("ns=%RU64\n", u64));
+ RTTESTI_CHECK_MSG((u64 = PDMAudioPropsFramesToNano(&Cfg441StereoS16, 255)) == 5782312, ("ns=%RU64\n", u64));
+ //RTTESTI_CHECK_MSG((u64 = DrvAudioHlpFramesToMicro(&Cfg441StereoS16, 44100)) == RT_US_1SEC,
+ // ("us=%RU64\n", u64));
+ RTTESTI_CHECK_MSG((u64 = PDMAudioPropsFramesToMilli(&Cfg441StereoS16, 44100)) == RT_MS_1SEC, ("ms=%RU64\n", u64));
+ RTTESTI_CHECK_MSG((u64 = PDMAudioPropsFramesToMilli(&Cfg441StereoS16, 255)) == 5, ("ms=%RU64\n", u64));
+
+ RTTESTI_CHECK_MSG((u32 = PDMAudioPropsNanoToFrames(&Cfg441StereoS16, RT_NS_1SEC)) == 44100, ("cb=%RU32\n", u32));
+ RTTESTI_CHECK_MSG((u32 = PDMAudioPropsNanoToFrames(&Cfg441StereoS16, 215876)) == 10, ("cb=%RU32\n", u32));
+ RTTESTI_CHECK_MSG((u32 = PDMAudioPropsMilliToFrames(&Cfg441StereoS16, RT_MS_1SEC)) == 44100, ("cb=%RU32\n", u32));
+ RTTESTI_CHECK_MSG((u32 = PDMAudioPropsMilliToFrames(&Cfg441StereoU32, 6)) == 265, ("cb=%RU32\n", u32));
+
+ RTTESTI_CHECK_MSG((u32 = PDMAudioPropsNanoToBytes(&Cfg441StereoS16, RT_NS_1SEC)) == 44100*2*2, ("cb=%RU32\n", u32));
+ RTTESTI_CHECK_MSG((u32 = PDMAudioPropsNanoToBytes(&Cfg441StereoS16, 702947)) == 31*2*2, ("cb=%RU32\n", u32));
+ RTTESTI_CHECK_MSG((u32 = PDMAudioPropsMilliToBytes(&Cfg441StereoS16, RT_MS_1SEC)) == 44100*2*2, ("cb=%RU32\n", u32));
+ RTTESTI_CHECK_MSG((u32 = PDMAudioPropsMilliToBytes(&Cfg441StereoS16, 5)) == 884, ("cb=%RU32\n", u32));
+
+ /* DrvAudioHlpClearBuf: */
+ uint8_t *pbPage;
+ int rc = RTTestGuardedAlloc(hTest, HOST_PAGE_SIZE, 0, false /*fHead*/, (void **)&pbPage);
+ RTTESTI_CHECK_RC_OK_RETV(rc);
+
+ memset(pbPage, 0x42, HOST_PAGE_SIZE);
+ PDMAudioPropsClearBuffer(&Cfg441StereoS16, pbPage, HOST_PAGE_SIZE, HOST_PAGE_SIZE / 4);
+ RTTESTI_CHECK(ASMMemIsZero(pbPage, HOST_PAGE_SIZE));
+
+ memset(pbPage, 0x42, HOST_PAGE_SIZE);
+ PDMAudioPropsClearBuffer(&Cfg441StereoU16, pbPage, HOST_PAGE_SIZE, HOST_PAGE_SIZE / 4);
+ for (uint32_t off = 0; off < HOST_PAGE_SIZE; off += 2)
+ RTTESTI_CHECK_MSG(pbPage[off] == 0 && pbPage[off + 1] == 0x80, ("off=%#x: %#x %x\n", off, pbPage[off], pbPage[off + 1]));
+
+ memset(pbPage, 0x42, HOST_PAGE_SIZE);
+ PDMAudioPropsClearBuffer(&Cfg441StereoU32, pbPage, HOST_PAGE_SIZE, HOST_PAGE_SIZE / 8);
+ for (uint32_t off = 0; off < HOST_PAGE_SIZE; off += 4)
+ RTTESTI_CHECK(pbPage[off] == 0 && pbPage[off + 1] == 0 && pbPage[off + 2] == 0 && pbPage[off + 3] == 0x80);
+
+
+ RTTestDisableAssertions(hTest);
+ memset(pbPage, 0x42, HOST_PAGE_SIZE);
+ PDMAudioPropsClearBuffer(&Cfg441StereoS16, pbPage, HOST_PAGE_SIZE, HOST_PAGE_SIZE); /* should adjust down the frame count. */
+ RTTESTI_CHECK(ASMMemIsZero(pbPage, HOST_PAGE_SIZE));
+
+ memset(pbPage, 0x42, HOST_PAGE_SIZE);
+ PDMAudioPropsClearBuffer(&Cfg441StereoU16, pbPage, HOST_PAGE_SIZE, HOST_PAGE_SIZE); /* should adjust down the frame count. */
+ for (uint32_t off = 0; off < HOST_PAGE_SIZE; off += 2)
+ RTTESTI_CHECK_MSG(pbPage[off] == 0 && pbPage[off + 1] == 0x80, ("off=%#x: %#x %x\n", off, pbPage[off], pbPage[off + 1]));
+
+ memset(pbPage, 0x42, HOST_PAGE_SIZE);
+ PDMAudioPropsClearBuffer(&Cfg441StereoU32, pbPage, HOST_PAGE_SIZE, HOST_PAGE_SIZE); /* should adjust down the frame count. */
+ for (uint32_t off = 0; off < HOST_PAGE_SIZE; off += 4)
+ RTTESTI_CHECK(pbPage[off] == 0 && pbPage[off + 1] == 0 && pbPage[off + 2] == 0 && pbPage[off + 3] == 0x80);
+ RTTestRestoreAssertions(hTest);
+
+ RTTestGuardedFree(hTest, pbPage);
+}
+
+
+static void tstSimple(RTTEST hTest)
+{
+ RTTestSub(hTest, "Simple");
+
+ /* 44100Hz, 2 Channels, S16 */
+ PDMAUDIOPCMPROPS config = PDMAUDIOPCMPROPS_INITIALIZER(
+ 2, /* Bytes */
+ true, /* Signed */
+ 2, /* Channels */
+ 44100, /* Hz */
+ false /* Swap Endian */
+ );
+
+ RTTESTI_CHECK(AudioHlpPcmPropsAreValidAndSupported(&config));
+
+ uint32_t cBufSize = _1K;
+
+ /*
+ * General stuff.
+ */
+ AUDIOMIXBUF mb;
+ RTTESTI_CHECK_RC_OK_RETV(AudioMixBufInit(&mb, "Single", &config, cBufSize));
+ RTTESTI_CHECK(AudioMixBufSize(&mb) == cBufSize);
+ RTTESTI_CHECK(AUDIOMIXBUF_B2F(&mb, AudioMixBufSizeBytes(&mb)) == cBufSize);
+ RTTESTI_CHECK(AUDIOMIXBUF_F2B(&mb, AudioMixBufSize(&mb)) == AudioMixBufSizeBytes(&mb));
+ RTTESTI_CHECK(AudioMixBufFree(&mb) == cBufSize);
+ RTTESTI_CHECK(AUDIOMIXBUF_F2B(&mb, AudioMixBufFree(&mb)) == AudioMixBufFreeBytes(&mb));
+
+ AUDIOMIXBUFWRITESTATE WriteState;
+ RTTESTI_CHECK_RC(AudioMixBufInitWriteState(&mb, &WriteState, &config), VINF_SUCCESS);
+
+ AUDIOMIXBUFPEEKSTATE PeekState;
+ RTTESTI_CHECK_RC(AudioMixBufInitPeekState(&mb, &PeekState, &config), VINF_SUCCESS);
+
+ /*
+ * A few writes (used to be the weird absolute writes).
+ */
+ uint32_t cFramesRead = 0, cFramesWritten = 0, cFramesWrittenAbs = 0;
+ int16_t aFrames16[2] = { 0xAA, 0xBB };
+ int32_t aFrames32[2] = { 0xCC, 0xDD };
+
+ RTTESTI_CHECK(AudioMixBufUsed(&mb) == 0);
+
+ AudioMixBufWrite(&mb, &WriteState, &aFrames16, sizeof(aFrames16), 0 /*offDstFrame*/, cBufSize / 4, &cFramesWritten);
+ RTTESTI_CHECK(cFramesWritten == 1 /* Frames */);
+ RTTESTI_CHECK(AudioMixBufUsed(&mb) == 0);
+ AudioMixBufCommit(&mb, cFramesWritten);
+ RTTESTI_CHECK(AudioMixBufUsed(&mb) == 1);
+ RTTESTI_CHECK(AudioMixBufReadPos(&mb) == 0);
+ RTTESTI_CHECK(AudioMixBufWritePos(&mb) == 1);
+
+ AudioMixBufWrite(&mb, &WriteState, &aFrames32, sizeof(aFrames32), 0 /*offDstFrame*/, cBufSize / 4, &cFramesWritten);
+ RTTESTI_CHECK(cFramesWritten == 2 /* Frames */);
+ AudioMixBufCommit(&mb, cFramesWritten);
+ RTTESTI_CHECK(AudioMixBufUsed(&mb) == 3);
+ RTTESTI_CHECK(AudioMixBufReadPos(&mb) == 0);
+ RTTESTI_CHECK(AudioMixBufWritePos(&mb) == 3);
+
+ /* Pretend we read the frames.*/
+ AudioMixBufAdvance(&mb, 3);
+ RTTESTI_CHECK(AudioMixBufUsed(&mb) == 0);
+ RTTESTI_CHECK(AudioMixBufReadPos(&mb) == 3);
+ RTTESTI_CHECK(AudioMixBufWritePos(&mb) == 3);
+
+ /* Fill up the buffer completely and check wraps. */
+
+ uint32_t cbSamples = PDMAudioPropsFramesToBytes(&config, cBufSize);
+ uint16_t *paSamples = (uint16_t *)RTMemAlloc(cbSamples);
+ RTTESTI_CHECK_RETV(paSamples);
+ AudioMixBufWrite(&mb, &WriteState, paSamples, cbSamples, 0 /*offDstFrame*/, cBufSize, &cFramesWritten);
+ RTTESTI_CHECK(cFramesWritten == cBufSize);
+ AudioMixBufCommit(&mb, cFramesWritten);
+ RTTESTI_CHECK(AudioMixBufUsed(&mb) == cBufSize);
+ RTTESTI_CHECK(AudioMixBufReadPos(&mb) == 3);
+ RTTESTI_CHECK(AudioMixBufWritePos(&mb) == 3);
+ RTMemFree(paSamples);
+ cbSamples = 0;
+
+ /*
+ * Writes and reads (used to be circular).
+ */
+ AudioMixBufDrop(&mb);
+
+ cFramesWrittenAbs = AudioMixBufUsed(&mb);
+
+ uint32_t cToWrite = AudioMixBufSize(&mb) - cFramesWrittenAbs - 1; /* -1 as padding plus -2 frames for above. */
+ for (uint32_t i = 0; i < cToWrite; i++)
+ {
+ AudioMixBufWrite(&mb, &WriteState, &aFrames16[0], sizeof(aFrames16), 0 /*offDstFrame*/, 1, &cFramesWritten);
+ RTTESTI_CHECK(cFramesWritten == 1);
+ AudioMixBufCommit(&mb, cFramesWritten);
+ }
+ RTTESTI_CHECK(!AudioMixBufIsEmpty(&mb));
+ RTTESTI_CHECK(AudioMixBufFree(&mb) == 1);
+ RTTESTI_CHECK(AudioMixBufFreeBytes(&mb) == AUDIOMIXBUF_F2B(&mb, 1U));
+ RTTESTI_CHECK(AudioMixBufUsed(&mb) == cToWrite + cFramesWrittenAbs /* + last absolute write */);
+
+ AudioMixBufWrite(&mb, &WriteState, &aFrames16[0], sizeof(aFrames16), 0 /*offDstFrame*/, 1, &cFramesWritten);
+ RTTESTI_CHECK(cFramesWritten == 1);
+ AudioMixBufCommit(&mb, cFramesWritten);
+ RTTESTI_CHECK(AudioMixBufFree(&mb) == 0);
+ RTTESTI_CHECK(AudioMixBufFreeBytes(&mb) == AUDIOMIXBUF_F2B(&mb, 0U));
+ RTTESTI_CHECK(AudioMixBufUsed(&mb) == cBufSize);
+
+ /* Reads. */
+ RTTESTI_CHECK(AudioMixBufReadPos(&mb) == 0);
+ uint32_t cbRead;
+ uint16_t aFrames16Buf[RT_ELEMENTS(aFrames16)];
+ uint32_t cToRead = AudioMixBufSize(&mb) - cFramesWrittenAbs - 1;
+ for (uint32_t i = 0; i < cToRead; i++)
+ {
+ AudioMixBufPeek(&mb, 0 /*offSrcFrame*/, 1, &cFramesRead, &PeekState, aFrames16Buf, sizeof(aFrames16Buf), &cbRead);
+ RTTESTI_CHECK(cFramesRead == 1);
+ RTTESTI_CHECK(cbRead == sizeof(aFrames16Buf));
+ AudioMixBufAdvance(&mb, cFramesRead);
+ RTTESTI_CHECK(AudioMixBufReadPos(&mb) == i + 1);
+ }
+ RTTESTI_CHECK(!AudioMixBufIsEmpty(&mb));
+ RTTESTI_CHECK(AudioMixBufFree(&mb) == AudioMixBufSize(&mb) - cFramesWrittenAbs - 1);
+ RTTESTI_CHECK(AudioMixBufFreeBytes(&mb) == AUDIOMIXBUF_F2B(&mb, cBufSize - cFramesWrittenAbs - 1));
+ RTTESTI_CHECK(AudioMixBufUsed(&mb) == cBufSize - cToRead);
+
+ AudioMixBufPeek(&mb, 0 /*offSrcFrame*/, 1, &cFramesRead, &PeekState, aFrames16Buf, sizeof(aFrames16Buf), &cbRead);
+ RTTESTI_CHECK(cFramesRead == 1);
+ RTTESTI_CHECK(cbRead == sizeof(aFrames16Buf));
+ AudioMixBufAdvance(&mb, cFramesRead);
+ RTTESTI_CHECK(AudioMixBufFree(&mb) == cBufSize - cFramesWrittenAbs);
+ RTTESTI_CHECK(AudioMixBufFreeBytes(&mb) == AUDIOMIXBUF_F2B(&mb, cBufSize - cFramesWrittenAbs));
+ RTTESTI_CHECK(AudioMixBufUsed(&mb) == cFramesWrittenAbs);
+ RTTESTI_CHECK(AudioMixBufReadPos(&mb) == 0);
+
+ AudioMixBufTerm(&mb);
+}
+
+/** @name Eight test samples represented in all basic formats.
+ * @{ */
+static uint8_t const g_au8TestSamples[8] = { 0x1, 0x11, 0x32, 0x7f, 0x80, 0x81, 0xbe, 0xff };
+static int8_t const g_ai8TestSamples[8] = { -127, -111, -78, -1, 0, 1, 62, 127 };
+static uint16_t const g_au16TestSamples[8] = { 0x100, 0x1100, 0x3200, 0x7f00, 0x8000, 0x8100, 0xbe00, 0xff00 };
+static int16_t const g_ai16TestSamples[8] = { -32512, -28416, -19968, -256, 0, 256, 15872, 32512 };
+static uint32_t const g_au32TestSamples[8] = { 0x1000000, 0x11000000, 0x32000000, 0x7f000000, 0x80000000, 0x81000000, 0xbe000000, 0xff000000 };
+static int32_t const g_ai32TestSamples[8] = { -2130706432, -1862270976, -1308622848, -16777216, 0, 16777216, 1040187392, 2130706432 };
+static int64_t const g_ai64TestSamples[8] = { -2130706432, -1862270976, -1308622848, -16777216, 0, 16777216, 1040187392, 2130706432 };
+static struct { void const *apv[2]; uint32_t cb; } g_aTestSamples[] =
+{
+ /* 0/0: */ { { NULL, NULL }, 0 },
+ /* 1/8: */ { { g_au8TestSamples, g_ai8TestSamples }, sizeof( g_au8TestSamples) },
+ /* 2/16: */ { { g_au16TestSamples, g_ai16TestSamples }, sizeof(g_au16TestSamples) },
+ /* 3/24: */ { { NULL, NULL }, 0 },
+ /* 4/32: */ { { g_au32TestSamples, g_ai32TestSamples }, sizeof(g_au32TestSamples) },
+ /* 5: */ { { NULL, NULL }, 0 },
+ /* 6: */ { { NULL, NULL }, 0 },
+ /* 7: */ { { NULL, NULL }, 0 },
+ /* 8:64 */ { { NULL, g_ai64TestSamples }, sizeof(g_ai64TestSamples) }, /* raw */
+};
+/** @} */
+
+/** Fills a buffer with samples from an g_aTestSamples entry. */
+static uint32_t tstFillBuf(PCPDMAUDIOPCMPROPS pCfg, void const *pvTestSamples, uint32_t iTestSample,
+ uint8_t *pbBuf, uint32_t cFrames)
+{
+ uint8_t const cTestSamples = RT_ELEMENTS(g_au8TestSamples);
+
+ cFrames *= PDMAudioPropsChannels(pCfg);
+ switch (PDMAudioPropsSampleSize(pCfg))
+ {
+ case 1:
+ {
+ uint8_t const * const pau8TestSamples = (uint8_t const *)pvTestSamples;
+ uint8_t *pu8Dst = (uint8_t *)pbBuf;
+ while (cFrames-- > 0)
+ {
+ *pu8Dst++ = pau8TestSamples[iTestSample];
+ iTestSample = (iTestSample + 1) % cTestSamples;
+ }
+ break;
+ }
+
+ case 2:
+ {
+ uint16_t const * const pau16TestSamples = (uint16_t const *)pvTestSamples;
+ uint16_t *pu16Dst = (uint16_t *)pbBuf;
+ while (cFrames-- > 0)
+ {
+ *pu16Dst++ = pau16TestSamples[iTestSample];
+ iTestSample = (iTestSample + 1) % cTestSamples;
+ }
+ break;
+ }
+
+ case 4:
+ {
+ uint32_t const * const pau32TestSamples = (uint32_t const *)pvTestSamples;
+ uint32_t *pu32Dst = (uint32_t *)pbBuf;
+ while (cFrames-- > 0)
+ {
+ *pu32Dst++ = pau32TestSamples[iTestSample];
+ iTestSample = (iTestSample + 1) % cTestSamples;
+ }
+ break;
+ }
+
+ case 8:
+ {
+ uint64_t const * const pau64TestSamples = (uint64_t const *)pvTestSamples;
+ uint64_t *pu64Dst = (uint64_t *)pbBuf;
+ while (cFrames-- > 0)
+ {
+ *pu64Dst++ = pau64TestSamples[iTestSample];
+ iTestSample = (iTestSample + 1) % cTestSamples;
+ }
+ break;
+ }
+
+ default:
+ AssertFailedBreak();
+ }
+ return iTestSample;
+}
+
+
+static void tstConversion(RTTEST hTest, uint8_t cSrcBits, bool fSrcSigned, uint8_t cSrcChs,
+ uint8_t cDstBits, bool fDstSigned, uint8_t cDstChs)
+{
+ RTTestSubF(hTest, "Conv %uch %c%u to %uch %c%u", cSrcChs, fSrcSigned ? 'S' : 'U', cSrcBits,
+ cDstChs, fDstSigned ? 'S' : 'U', cDstBits);
+
+ PDMAUDIOPCMPROPS CfgSrc, CfgDst;
+ PDMAudioPropsInitEx(&CfgSrc, cSrcBits / 8, fSrcSigned, cSrcChs, 44100, g_fLittleEndian, cSrcBits == 64 /*fRaw*/);
+ PDMAudioPropsInitEx(&CfgDst, cDstBits / 8, fDstSigned, cDstChs, 44100, g_fLittleEndian, cDstBits == 64 /*fRaw*/);
+
+ void const * const pvSrcTestSamples = g_aTestSamples[cSrcBits / 8].apv[fSrcSigned];
+ void const * const pvDstTestSamples = g_aTestSamples[cDstBits / 8].apv[fDstSigned];
+ uint32_t const cMixBufFrames = RTRandU32Ex(128, 16384);
+ uint32_t const cIterations = RTRandU32Ex(256, 1536);
+ uint32_t const cbSrcBuf = PDMAudioPropsFramesToBytes(&CfgSrc, cMixBufFrames + 64);
+ uint8_t * const pbSrcBuf = (uint8_t *)RTMemAllocZ(cbSrcBuf);
+ uint32_t const cbDstBuf = PDMAudioPropsFramesToBytes(&CfgDst, cMixBufFrames + 64);
+ uint8_t * const pbDstBuf = (uint8_t *)RTMemAllocZ(cbDstBuf);
+ uint8_t * const pbDstExpect = (uint8_t *)RTMemAllocZ(cbDstBuf);
+ RTTESTI_CHECK_RETV(pbSrcBuf);
+ RTTESTI_CHECK_RETV(pbDstBuf);
+ RTTESTI_CHECK_RETV(pbDstExpect);
+
+ AUDIOMIXBUF MixBuf;
+ RTTESTI_CHECK_RC_RETV(AudioMixBufInit(&MixBuf, "FormatOutputConversion", &CfgSrc, cMixBufFrames), VINF_SUCCESS);
+ AUDIOMIXBUFWRITESTATE WriteState;
+ RTTESTI_CHECK_RC_RETV(AudioMixBufInitWriteState(&MixBuf, &WriteState, &CfgSrc), VINF_SUCCESS);
+ AUDIOMIXBUFWRITESTATE WriteStateIgnZero = WriteState; RT_NOREF(WriteStateIgnZero);
+ AUDIOMIXBUFPEEKSTATE PeekState;
+ RTTESTI_CHECK_RC_RETV(AudioMixBufInitPeekState(&MixBuf, &PeekState, &CfgDst), VINF_SUCCESS);
+
+ uint32_t iSrcTestSample = 0;
+ uint32_t iDstTestSample = 0;
+ //RTTestPrintf(hTest, RTTESTLVL_ALWAYS, "cIterations=%u\n", cIterations);
+ for (uint32_t iIteration = 0; iIteration < cIterations; iIteration++)
+ {
+ /* Write some frames to the buffer. */
+ uint32_t const cSrcFramesToWrite = iIteration < 16 ? iIteration + 1
+ : AudioMixBufFree(&MixBuf) ? RTRandU32Ex(1, AudioMixBufFree(&MixBuf)) : 0;
+ if (cSrcFramesToWrite > 0)
+ {
+ uint32_t const cbSrcToWrite = PDMAudioPropsFramesToBytes(&CfgSrc, cSrcFramesToWrite);
+ uint32_t cFrames = RTRandU32();
+ switch (RTRandU32Ex(0, 3))
+ {
+ default:
+ iSrcTestSample = tstFillBuf(&CfgSrc, pvSrcTestSamples, iSrcTestSample, pbSrcBuf, cSrcFramesToWrite);
+ AudioMixBufWrite(&MixBuf, &WriteState, pbSrcBuf, cbSrcToWrite, 0 /*offDstFrame*/, cSrcFramesToWrite, &cFrames);
+ RTTESTI_CHECK(cFrames == cSrcFramesToWrite);
+ break;
+
+ case 1: /* zero & blend */
+ AudioMixBufSilence(&MixBuf, &WriteStateIgnZero, 0 /*offFrame*/, cSrcFramesToWrite);
+ iSrcTestSample = tstFillBuf(&CfgSrc, pvSrcTestSamples, iSrcTestSample, pbSrcBuf, cSrcFramesToWrite);
+ AudioMixBufBlend(&MixBuf, &WriteState, pbSrcBuf, cbSrcToWrite, 0 /*offDstFrame*/, cSrcFramesToWrite, &cFrames);
+ RTTESTI_CHECK(cFrames == cSrcFramesToWrite);
+ break;
+
+ case 2: /* blend same equal data twice */
+ {
+ AUDIOMIXBUFWRITESTATE WriteStateSame = WriteState;
+ iSrcTestSample = tstFillBuf(&CfgSrc, pvSrcTestSamples, iSrcTestSample, pbSrcBuf, cSrcFramesToWrite);
+ AudioMixBufWrite(&MixBuf, &WriteState, pbSrcBuf, cbSrcToWrite, 0 /*offDstFrame*/, cSrcFramesToWrite, &cFrames);
+ RTTESTI_CHECK(cFrames == cSrcFramesToWrite);
+ AudioMixBufBlend(&MixBuf, &WriteStateSame, pbSrcBuf, cbSrcToWrite, 0 /*offDstFrame*/, cSrcFramesToWrite, &cFrames);
+ RTTESTI_CHECK(cFrames == cSrcFramesToWrite);
+ break;
+ }
+ case 3: /* write & blend with zero */
+ {
+ AUDIOMIXBUFWRITESTATE WriteStateSame = WriteState;
+ iSrcTestSample = tstFillBuf(&CfgSrc, pvSrcTestSamples, iSrcTestSample, pbSrcBuf, cSrcFramesToWrite);
+ AudioMixBufWrite(&MixBuf, &WriteState, pbSrcBuf, cbSrcToWrite, 0 /*offDstFrame*/, cSrcFramesToWrite, &cFrames);
+ RTTESTI_CHECK(cFrames == cSrcFramesToWrite);
+ PDMAudioPropsClearBuffer(&CfgSrc, pbSrcBuf, cbSrcToWrite, cSrcFramesToWrite);
+ AudioMixBufBlend(&MixBuf, &WriteStateSame, pbSrcBuf, cbSrcToWrite, 0 /*offDstFrame*/, cSrcFramesToWrite, &cFrames);
+ RTTESTI_CHECK(cFrames == cSrcFramesToWrite);
+ break;
+ }
+ }
+ AudioMixBufCommit(&MixBuf, cSrcFramesToWrite);
+ }
+
+ /* Read some frames back. */
+ uint32_t const cUsed = AudioMixBufUsed(&MixBuf);
+ uint32_t const cDstFramesToRead = iIteration < 16 ? iIteration + 1 : iIteration + 5 >= cIterations ? cUsed
+ : cUsed ? RTRandU32Ex(1, cUsed) : 0;
+ if (cDstFramesToRead > 0)
+ {
+ uint32_t const cbDstToRead = PDMAudioPropsFramesToBytes(&CfgDst, cDstFramesToRead);
+ uint32_t cbRead = RTRandU32();
+ uint32_t cFrames = RTRandU32();
+ RTRandBytes(pbDstBuf, cbDstToRead);
+ AudioMixBufPeek(&MixBuf, 0 /*offSrcFrame*/, (iIteration & 3) != 2 ? cDstFramesToRead : cUsed, &cFrames,
+ &PeekState, pbDstBuf, (iIteration & 3) != 3 ? cbDstToRead : cbDstBuf, &cbRead);
+ RTTESTI_CHECK(cFrames == cDstFramesToRead);
+ RTTESTI_CHECK(cbRead == cbDstToRead);
+ AudioMixBufAdvance(&MixBuf, cFrames);
+
+ /* Verify if we can. */
+ if (PDMAudioPropsChannels(&CfgSrc) == PDMAudioPropsChannels(&CfgDst))
+ {
+ iDstTestSample = tstFillBuf(&CfgDst, pvDstTestSamples, iDstTestSample, pbDstExpect, cFrames);
+ if (memcmp(pbDstExpect, pbDstBuf, cbRead) == 0)
+ { /* likely */ }
+ else
+ {
+ RTTestFailed(hTest,
+ "mismatch: %.*Rhxs\n"
+ "expected: %.*Rhxs\n"
+ "iIteration=%u cDstFramesToRead=%u cbRead=%#x\n",
+ RT_MIN(cbRead, 48), pbDstBuf,
+ RT_MIN(cbRead, 48), pbDstExpect,
+ iIteration, cDstFramesToRead, cbRead);
+ break;
+ }
+ }
+ }
+ }
+
+ AudioMixBufTerm(&MixBuf);
+ RTMemFree(pbSrcBuf);
+ RTMemFree(pbDstBuf);
+ RTMemFree(pbDstExpect);
+}
+
+
+#if 0 /** @todo rewrite to non-parent/child setup */
+static void tstDownsampling(RTTEST hTest, uint32_t uFromHz, uint32_t uToHz)
+{
+ RTTestSubF(hTest, "Downsampling %u to %u Hz (S16)", uFromHz, uToHz);
+
+ struct { int16_t l, r; }
+ aSrcFrames[4096],
+ aDstFrames[4096];
+
+ /* Parent (destination) buffer is xxxHz 2ch S16 */
+ uint32_t const cFramesParent = RTRandU32Ex(16, RT_ELEMENTS(aDstFrames));
+ PDMAUDIOPCMPROPS const CfgDst = PDMAUDIOPCMPROPS_INITIALIZER(2 /*cbSample*/, true /*fSigned*/, 2 /*ch*/, uToHz, false /*fSwap*/);
+ RTTESTI_CHECK(AudioHlpPcmPropsAreValid(&CfgDst));
+ AUDIOMIXBUF Parent;
+ RTTESTI_CHECK_RC_OK_RETV(AudioMixBufInit(&Parent, "ParentDownsampling", &CfgDst, cFramesParent));
+
+ /* Child (source) buffer is yyykHz 2ch S16 */
+ PDMAUDIOPCMPROPS const CfgSrc = PDMAUDIOPCMPROPS_INITIALIZER(2 /*cbSample*/, true /*fSigned*/, 2 /*ch*/, uFromHz, false /*fSwap*/);
+ RTTESTI_CHECK(AudioHlpPcmPropsAreValid(&CfgSrc));
+ uint32_t const cFramesChild = RTRandU32Ex(32, RT_ELEMENTS(aSrcFrames));
+ AUDIOMIXBUF Child;
+ RTTESTI_CHECK_RC_OK_RETV(AudioMixBufInit(&Child, "ChildDownsampling", &CfgSrc, cFramesChild));
+ RTTESTI_CHECK_RC_OK_RETV(AudioMixBufLinkTo(&Child, &Parent));
+
+ /*
+ * Test parameters.
+ */
+ uint32_t const cMaxSrcFrames = RT_MIN(cFramesParent * uFromHz / uToHz - 1, cFramesChild);
+ uint32_t const cIterations = RTRandU32Ex(4, 128);
+ RTTestErrContext(hTest, "cFramesParent=%RU32 cFramesChild=%RU32 cMaxSrcFrames=%RU32 cIterations=%RU32",
+ cFramesParent, cFramesChild, cMaxSrcFrames, cIterations);
+ RTTestPrintf(hTest, RTTESTLVL_DEBUG, "cFramesParent=%RU32 cFramesChild=%RU32 cMaxSrcFrames=%RU32 cIterations=%RU32\n",
+ cFramesParent, cFramesChild, cMaxSrcFrames, cIterations);
+
+ /*
+ * We generate a simple "A" sine wave as input.
+ */
+ uint32_t iSrcFrame = 0;
+ uint32_t iDstFrame = 0;
+ double rdFixed = 2.0 * M_PI * 440.0 /* A */ / PDMAudioPropsHz(&CfgSrc); /* Fixed sin() input. */
+ for (uint32_t i = 0; i < cIterations; i++)
+ {
+ RTTestPrintf(hTest, RTTESTLVL_DEBUG, "i=%RU32\n", i);
+
+ /*
+ * Generate source frames and write them.
+ */
+ uint32_t const cSrcFrames = i < cIterations / 2
+ ? RTRandU32Ex(2, cMaxSrcFrames) & ~(uint32_t)1
+ : RTRandU32Ex(1, cMaxSrcFrames - 1) | 1;
+ for (uint32_t j = 0; j < cSrcFrames; j++, iSrcFrame++)
+ aSrcFrames[j].r = aSrcFrames[j].l = 32760 /*Amplitude*/ * sin(rdFixed * iSrcFrame);
+
+ uint32_t cSrcFramesWritten = UINT32_MAX / 2;
+ RTTESTI_CHECK_RC_OK_BREAK(AudioMixBufWriteAt(&Child, 0, &aSrcFrames, cSrcFrames * sizeof(aSrcFrames[0]),
+ &cSrcFramesWritten));
+ RTTESTI_CHECK_MSG_BREAK(cSrcFrames == cSrcFramesWritten,
+ ("cSrcFrames=%RU32 vs cSrcFramesWritten=%RU32\n", cSrcFrames, cSrcFramesWritten));
+
+ /*
+ * Mix them.
+ */
+ uint32_t cSrcFramesMixed = UINT32_MAX / 2;
+ RTTESTI_CHECK_RC_OK_BREAK(AudioMixBufMixToParent(&Child, cSrcFramesWritten, &cSrcFramesMixed));
+ RTTESTI_CHECK_MSG(AudioMixBufUsed(&Child) == 0, ("%RU32\n", AudioMixBufUsed(&Child)));
+ RTTESTI_CHECK_MSG_BREAK(cSrcFramesWritten == cSrcFramesMixed,
+ ("cSrcFramesWritten=%RU32 cSrcFramesMixed=%RU32\n", cSrcFramesWritten, cSrcFramesMixed));
+ RTTESTI_CHECK_MSG_BREAK(AudioMixBufUsed(&Child) == 0, ("%RU32\n", AudioMixBufUsed(&Child)));
+
+ /*
+ * Read out the parent buffer.
+ */
+ uint32_t cDstFrames = AudioMixBufUsed(&Parent);
+ while (cDstFrames > 0)
+ {
+ uint32_t cFramesRead = UINT32_MAX / 2;
+ RTTESTI_CHECK_RC_OK_BREAK(AudioMixBufAcquireReadBlock(&Parent, aDstFrames, sizeof(aDstFrames), &cFramesRead));
+ RTTESTI_CHECK_MSG(cFramesRead > 0 && cFramesRead <= cDstFrames,
+ ("cFramesRead=%RU32 cDstFrames=%RU32\n", cFramesRead, cDstFrames));
+
+ AudioMixBufReleaseReadBlock(&Parent, cFramesRead);
+ AudioMixBufFinish(&Parent, cFramesRead);
+
+ iDstFrame += cFramesRead;
+ cDstFrames -= cFramesRead;
+ RTTESTI_CHECK(AudioMixBufUsed(&Parent) == cDstFrames);
+ }
+ }
+
+ RTTESTI_CHECK(AudioMixBufUsed(&Parent) == 0);
+ RTTESTI_CHECK(AudioMixBufLive(&Child) == 0);
+ uint32_t const cDstMinExpect = (uint64_t)iSrcFrame * uToHz / uFromHz;
+ uint32_t const cDstMaxExpect = ((uint64_t)iSrcFrame * uToHz + uFromHz - 1) / uFromHz;
+ RTTESTI_CHECK_MSG(iDstFrame == cDstMinExpect || iDstFrame == cDstMaxExpect,
+ ("iSrcFrame=%#x -> %#x,%#x; iDstFrame=%#x\n", iSrcFrame, cDstMinExpect, cDstMaxExpect, iDstFrame));
+
+ AudioMixBufDestroy(&Parent);
+ AudioMixBufDestroy(&Child);
+}
+#endif
+
+
+static void tstNewPeek(RTTEST hTest, uint32_t uFromHz, uint32_t uToHz)
+{
+ RTTestSubF(hTest, "New peek %u to %u Hz (S16)", uFromHz, uToHz);
+
+ struct { int16_t l, r; }
+ aSrcFrames[4096],
+ aDstFrames[4096];
+
+ /* Mix buffer is uFromHz 2ch S16 */
+ uint32_t const cFrames = RTRandU32Ex(16, RT_ELEMENTS(aSrcFrames));
+ PDMAUDIOPCMPROPS const CfgSrc = PDMAUDIOPCMPROPS_INITIALIZER(2 /*cbSample*/, true /*fSigned*/, 2 /*ch*/, uFromHz, false /*fSwap*/);
+ RTTESTI_CHECK(AudioHlpPcmPropsAreValidAndSupported(&CfgSrc));
+ AUDIOMIXBUF MixBuf;
+ RTTESTI_CHECK_RC_OK_RETV(AudioMixBufInit(&MixBuf, "NewPeekMixBuf", &CfgSrc, cFrames));
+
+ /* Write state (source). */
+ AUDIOMIXBUFWRITESTATE WriteState;
+ RTTESTI_CHECK_RC_OK_RETV(AudioMixBufInitWriteState(&MixBuf, &WriteState, &CfgSrc));
+
+ /* Peek state (destination) is uToHz 2ch S16 */
+ PDMAUDIOPCMPROPS const CfgDst = PDMAUDIOPCMPROPS_INITIALIZER(2 /*cbSample*/, true /*fSigned*/, 2 /*ch*/, uToHz, false /*fSwap*/);
+ RTTESTI_CHECK(AudioHlpPcmPropsAreValidAndSupported(&CfgDst));
+ AUDIOMIXBUFPEEKSTATE PeekState;
+ RTTESTI_CHECK_RC_OK_RETV(AudioMixBufInitPeekState(&MixBuf, &PeekState, &CfgDst));
+
+ /*
+ * Test parameters.
+ */
+ uint32_t const cMaxSrcFrames = RT_MIN(cFrames * uFromHz / uToHz - 1, cFrames);
+ uint32_t const cIterations = RTRandU32Ex(64, 1024);
+ RTTestErrContext(hTest, "cFrames=%RU32 cMaxSrcFrames=%RU32 cIterations=%RU32", cFrames, cMaxSrcFrames, cIterations);
+ RTTestPrintf(hTest, RTTESTLVL_DEBUG, "cFrames=%RU32 cMaxSrcFrames=%RU32 cIterations=%RU32\n",
+ cFrames, cMaxSrcFrames, cIterations);
+
+ /*
+ * We generate a simple "A" sine wave as input.
+ */
+ uint32_t iSrcFrame = 0;
+ uint32_t iDstFrame = 0;
+ double rdFixed = 2.0 * M_PI * 440.0 /* A */ / PDMAudioPropsHz(&CfgSrc); /* Fixed sin() input. */
+ for (uint32_t i = 0; i < cIterations; i++)
+ {
+ RTTestPrintf(hTest, RTTESTLVL_DEBUG, "i=%RU32\n", i);
+
+ /*
+ * Generate source frames and write them.
+ */
+ uint32_t const cSrcFrames = i < cIterations / 2
+ ? RTRandU32Ex(2, cMaxSrcFrames) & ~(uint32_t)1
+ : RTRandU32Ex(1, cMaxSrcFrames - 1) | 1;
+ for (uint32_t j = 0; j < cSrcFrames; j++, iSrcFrame++)
+ aSrcFrames[j].r = aSrcFrames[j].l = 32760 /*Amplitude*/ * sin(rdFixed * iSrcFrame);
+
+ uint32_t cSrcFramesWritten = UINT32_MAX / 2;
+ AudioMixBufWrite(&MixBuf, &WriteState, &aSrcFrames[0], cSrcFrames * sizeof(aSrcFrames[0]),
+ 0 /*offDstFrame*/, cSrcFrames, &cSrcFramesWritten);
+ RTTESTI_CHECK_MSG_BREAK(cSrcFrames == cSrcFramesWritten,
+ ("cSrcFrames=%RU32 vs cSrcFramesWritten=%RU32 cLiveFrames=%RU32\n",
+ cSrcFrames, cSrcFramesWritten, AudioMixBufUsed(&MixBuf)));
+ AudioMixBufCommit(&MixBuf, cSrcFrames);
+
+ /*
+ * Read out all the frames using the peek function.
+ */
+ uint32_t offSrcFrame = 0;
+ while (offSrcFrame < cSrcFramesWritten)
+ {
+ uint32_t cSrcFramesToRead = cSrcFramesWritten - offSrcFrame;
+ uint32_t cTmp = (uint64_t)cSrcFramesToRead * uToHz / uFromHz;
+ if (cTmp + 32 >= RT_ELEMENTS(aDstFrames))
+ cSrcFramesToRead = ((uint64_t)RT_ELEMENTS(aDstFrames) - 32) * uFromHz / uToHz; /* kludge */
+
+ uint32_t cSrcFramesPeeked = UINT32_MAX / 4;
+ uint32_t cbDstPeeked = UINT32_MAX / 2;
+ RTRandBytes(aDstFrames, sizeof(aDstFrames));
+ AudioMixBufPeek(&MixBuf, offSrcFrame, cSrcFramesToRead, &cSrcFramesPeeked,
+ &PeekState, aDstFrames, sizeof(aDstFrames), &cbDstPeeked);
+ uint32_t cDstFramesPeeked = PDMAudioPropsBytesToFrames(&CfgDst, cbDstPeeked);
+ RTTESTI_CHECK(cbDstPeeked > 0 || cSrcFramesPeeked > 0);
+
+ if (uFromHz == uToHz)
+ {
+ for (uint32_t iDst = 0; iDst < cDstFramesPeeked; iDst++)
+ if (memcmp(&aDstFrames[iDst], &aSrcFrames[offSrcFrame + iDst], sizeof(aSrcFrames[0])) != 0)
+ RTTestFailed(hTest, "Frame #%u differs: %#x / %#x, expected %#x / %#x\n", iDstFrame + iDst,
+ aDstFrames[iDst].l, aDstFrames[iDst].r,
+ aSrcFrames[iDst + offSrcFrame].l, aSrcFrames[iDst + offSrcFrame].r);
+ }
+
+ offSrcFrame += cSrcFramesPeeked;
+ iDstFrame += cDstFramesPeeked;
+ }
+
+ /*
+ * Then advance.
+ */
+ AudioMixBufAdvance(&MixBuf, cSrcFrames);
+ RTTESTI_CHECK(AudioMixBufUsed(&MixBuf) == 0);
+ }
+
+ /** @todo this is a bit lax... */
+ uint32_t const cDstMinExpect = ((uint64_t)iSrcFrame * uToHz - uFromHz - 1) / uFromHz;
+ uint32_t const cDstMaxExpect = ((uint64_t)iSrcFrame * uToHz + uFromHz - 1) / uFromHz;
+ RTTESTI_CHECK_MSG(iDstFrame >= cDstMinExpect && iDstFrame <= cDstMaxExpect,
+ ("iSrcFrame=%#x -> %#x..%#x; iDstFrame=%#x (delta %d)\n",
+ iSrcFrame, cDstMinExpect, cDstMaxExpect, iDstFrame, (cDstMinExpect + cDstMaxExpect) / 2 - iDstFrame));
+
+ AudioMixBufTerm(&MixBuf);
+}
+
+/* Test volume control. */
+static void tstVolume(RTTEST hTest)
+{
+ RTTestSub(hTest, "Volume control (44.1kHz S16 2ch)");
+ uint32_t const cBufSize = 256;
+
+ /*
+ * Configure a mixbuf where we read and write 44.1kHz S16 2ch.
+ */
+ PDMAUDIOPCMPROPS const Cfg = PDMAUDIOPCMPROPS_INITIALIZER(
+ 2, /* Bytes */
+ true, /* Signed */
+ 2, /* Channels */
+ 44100, /* Hz */
+ false /* Swap Endian */
+ );
+ AUDIOMIXBUF MixBuf;
+ RTTESTI_CHECK_RC_RETV(AudioMixBufInit(&MixBuf, "Volume", &Cfg, cBufSize), VINF_SUCCESS);
+
+ AUDIOMIXBUFWRITESTATE WriteState;
+ RTTESTI_CHECK_RC_RETV(AudioMixBufInitWriteState(&MixBuf, &WriteState, &Cfg), VINF_SUCCESS);
+
+ AUDIOMIXBUFPEEKSTATE PeekState;
+ RTTESTI_CHECK_RC_RETV(AudioMixBufInitPeekState(&MixBuf, &PeekState, &Cfg), VINF_SUCCESS);
+
+ /*
+ * A few 16-bit signed test samples.
+ */
+ static int16_t const s_aFrames16S[16] =
+ {
+ INT16_MIN, INT16_MIN + 1, -128, -64, -4, -1, 0, 1,
+ 2, 255, 256, INT16_MAX / 2, INT16_MAX - 2, INT16_MAX - 1, INT16_MAX, 0,
+ };
+
+ /*
+ * 1) Full volume/0dB attenuation (255).
+ */
+ PDMAUDIOVOLUME Vol = PDMAUDIOVOLUME_INITIALIZER_MAX;
+ AudioMixBufSetVolume(&MixBuf, &Vol);
+
+ /* Write all the test frames to the mixer buffer: */
+ uint32_t cFramesWritten;
+ AudioMixBufWrite(&MixBuf, &WriteState, &s_aFrames16S[0], sizeof(s_aFrames16S), 0 /*offDstFrame*/, cBufSize, &cFramesWritten);
+ RTTESTI_CHECK(cFramesWritten == RT_ELEMENTS(s_aFrames16S) / 2);
+ AudioMixBufCommit(&MixBuf, cFramesWritten);
+
+ /* Read them back. We should get them back just like we wrote them. */
+ uint16_t au16Buf[cBufSize * 2];
+ uint32_t cFramesPeeked;
+ uint32_t cbPeeked;
+ AudioMixBufPeek(&MixBuf, 0 /*offSrcFrame*/, cFramesWritten, &cFramesPeeked, &PeekState, au16Buf, sizeof(au16Buf), &cbPeeked);
+ RTTESTI_CHECK(cFramesPeeked == cFramesWritten);
+ RTTESTI_CHECK(cbPeeked == PDMAudioPropsFramesToBytes(&Cfg, cFramesPeeked));
+ AudioMixBufAdvance(&MixBuf, cFramesPeeked);
+
+ /* Check that at 0dB the frames came out unharmed. */
+ if (memcmp(au16Buf, s_aFrames16S, sizeof(s_aFrames16S)) != 0)
+ RTTestFailed(hTest,
+ "0dB test failed\n"
+ "mismatch: %.*Rhxs\n"
+ "expected: %.*Rhxs\n",
+ sizeof(s_aFrames16S), au16Buf, sizeof(s_aFrames16S), s_aFrames16S);
+
+ /*
+ * 2) Half volume/-6dB attenuation (16 steps down).
+ */
+ PDMAudioVolumeInitFromStereo(&Vol, false, 255 - 16, 255 - 16);
+ AudioMixBufSetVolume(&MixBuf, &Vol);
+
+ /* Write all the test frames to the mixer buffer: */
+ AudioMixBufWrite(&MixBuf, &WriteState, &s_aFrames16S[0], sizeof(s_aFrames16S), 0 /*offDstFrame*/, cBufSize, &cFramesWritten);
+ RTTESTI_CHECK(cFramesWritten == RT_ELEMENTS(s_aFrames16S) / 2);
+ AudioMixBufCommit(&MixBuf, cFramesWritten);
+
+ /* Read them back. We should get them back just like we wrote them. */
+ AudioMixBufPeek(&MixBuf, 0 /*offSrcFrame*/, cFramesWritten, &cFramesPeeked, &PeekState, au16Buf, sizeof(au16Buf), &cbPeeked);
+ RTTESTI_CHECK(cFramesPeeked == cFramesWritten);
+ RTTESTI_CHECK(cbPeeked == PDMAudioPropsFramesToBytes(&Cfg, cFramesPeeked));
+ AudioMixBufAdvance(&MixBuf, cFramesPeeked);
+
+ /* Check that at -6dB the sample values are halved. */
+ int16_t ai16Expect[sizeof(s_aFrames16S) / 2];
+ memcpy(ai16Expect, s_aFrames16S, sizeof(ai16Expect));
+ for (uintptr_t i = 0; i < RT_ELEMENTS(ai16Expect); i++)
+ ai16Expect[i] >>= 1; /* /= 2 - not the same for signed numbers; */
+ if (memcmp(au16Buf, ai16Expect, sizeof(ai16Expect)) != 0)
+ RTTestFailed(hTest,
+ "-6dB test failed\n"
+ "mismatch: %.*Rhxs\n"
+ "expected: %.*Rhxs\n"
+ "wrote: %.*Rhxs\n",
+ sizeof(ai16Expect), au16Buf, sizeof(ai16Expect), ai16Expect, sizeof(s_aFrames16S), s_aFrames16S);
+
+ AudioMixBufTerm(&MixBuf);
+}
+
+
+int main(int argc, char **argv)
+{
+ RTR3InitExe(argc, &argv, 0);
+
+ /*
+ * Initialize IPRT and create the test.
+ */
+ RTTEST hTest;
+ int rc = RTTestInitAndCreate("tstAudioMixBuffer", &hTest);
+ if (rc)
+ return rc;
+ RTTestBanner(hTest);
+
+ tstBasics(hTest);
+ tstSimple(hTest);
+
+ /* Run tstConversion for all combinations we have test data. */
+ for (unsigned iSrc = 0; iSrc < RT_ELEMENTS(g_aTestSamples); iSrc++)
+ {
+ for (unsigned iSrcSigned = 0; iSrcSigned < RT_ELEMENTS(g_aTestSamples[0].apv); iSrcSigned++)
+ if (g_aTestSamples[iSrc].apv[iSrcSigned])
+ for (unsigned cSrcChs = 1; cSrcChs <= 2; cSrcChs++)
+ for (unsigned iDst = 0; iDst < RT_ELEMENTS(g_aTestSamples); iDst++)
+ for (unsigned iDstSigned = 0; iDstSigned < RT_ELEMENTS(g_aTestSamples[0].apv); iDstSigned++)
+ if (g_aTestSamples[iDst].apv[iDstSigned])
+ for (unsigned cDstChs = 1; cDstChs <= 2; cDstChs++)
+ tstConversion(hTest, iSrc * 8, iSrcSigned == 1, cSrcChs,
+ /*->*/ iDst * 8, iDstSigned == 1, cDstChs);
+ }
+
+#if 0 /** @todo rewrite to non-parent/child setup */
+ tstDownsampling(hTest, 44100, 22050);
+ tstDownsampling(hTest, 48000, 44100);
+ tstDownsampling(hTest, 48000, 22050);
+ tstDownsampling(hTest, 48000, 11000);
+#endif
+ tstNewPeek(hTest, 48000, 48000);
+ tstNewPeek(hTest, 48000, 11000);
+ tstNewPeek(hTest, 48000, 44100);
+ tstNewPeek(hTest, 44100, 22050);
+ tstNewPeek(hTest, 44100, 11000);
+ //tstNewPeek(hTest, 11000, 48000);
+ //tstNewPeek(hTest, 22050, 44100);
+
+ tstVolume(hTest);
+
+ /*
+ * Summary
+ */
+ return RTTestSummaryAndDestroy(hTest);
+}
diff --git a/src/VBox/Devices/Audio/testcase/tstAudioTestService.cpp b/src/VBox/Devices/Audio/testcase/tstAudioTestService.cpp
new file mode 100644
index 00000000..9868218c
--- /dev/null
+++ b/src/VBox/Devices/Audio/testcase/tstAudioTestService.cpp
@@ -0,0 +1,178 @@
+/* $Id: tstAudioTestService.cpp $ */
+/** @file
+ * Audio testcase - Tests for the Audio Test Service (ATS).
+ */
+
+/*
+ * Copyright (C) 2021-2023 Oracle and/or its affiliates.
+ *
+ * This file is part of VirtualBox base platform packages, as
+ * available from https://www.virtualbox.org.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation, in version 3 of the
+ * License.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, see <https://www.gnu.org/licenses>.
+ *
+ * SPDX-License-Identifier: GPL-3.0-only
+ */
+
+
+/*********************************************************************************************************************************
+* Header Files *
+*********************************************************************************************************************************/
+
+#include <iprt/errcore.h>
+#include <iprt/file.h>
+#include <iprt/initterm.h>
+#include <iprt/mem.h>
+#include <iprt/rand.h>
+#include <iprt/stream.h>
+#include <iprt/string.h>
+#include <iprt/test.h>
+
+#include "../AudioTestService.h"
+#include "../AudioTestServiceClient.h"
+
+
+static size_t g_cToRead = _1M;
+static size_t g_cbRead = 0;
+
+
+/** @copydoc ATSCALLBACKS::pfnTestSetSendRead */
+static DECLCALLBACK(int) tstTestSetSendReadCallback(void const *pvUser,
+ const char *pszTag, void *pvBuf, size_t cbBuf, size_t *pcbRead)
+{
+ RT_NOREF(pvUser, pszTag);
+
+ size_t cbToRead = RT_MIN(g_cToRead - g_cbRead, cbBuf);
+ if (cbToRead)
+ {
+ memset(pvBuf, 0x42, cbToRead);
+ g_cbRead += cbToRead;
+ }
+
+ *pcbRead = cbToRead;
+
+ return VINF_SUCCESS;
+}
+
+int main(int argc, char **argv)
+{
+ RTR3InitExe(argc, &argv, 0);
+
+ /*
+ * Initialize IPRT and create the test.
+ */
+ RTTEST hTest;
+ int rc = RTTestInitAndCreate("tstAudioTestService", &hTest);
+ if (rc)
+ return rc;
+ RTTestBanner(hTest);
+
+ ATSCALLBACKS Callbacks;
+ RT_ZERO(Callbacks);
+ Callbacks.pfnTestSetSendRead = tstTestSetSendReadCallback;
+
+ ATSSERVER Srv;
+ rc = AudioTestSvcInit(&Srv, &Callbacks);
+ RTTEST_CHECK_RC_OK(hTest, rc);
+
+ RTGETOPTUNION Val;
+ RT_ZERO(Val);
+
+ Val.u32 = ATSCONNMODE_SERVER;
+ rc = AudioTestSvcHandleOption(&Srv, ATSTCPOPT_CONN_MODE, &Val);
+ RTTEST_CHECK_RC_OK(hTest, rc);
+
+ ATSCLIENT Client;
+
+ if (RT_SUCCESS(rc))
+ {
+ uint16_t uPort = ATS_TCP_DEF_BIND_PORT_HOST;
+
+ for (unsigned i = 0; i < 64; i++)
+ {
+ Val.u16 = uPort;
+ rc = AudioTestSvcHandleOption(&Srv, ATSTCPOPT_BIND_PORT, &Val);
+ RTTEST_CHECK_RC_OK(hTest, rc);
+
+ rc = AudioTestSvcStart(&Srv);
+ if (RT_SUCCESS(rc))
+ break;
+
+ RTTestPrintf(hTest, RTTESTLVL_ALWAYS, "Port %RU32 already used\n", uPort);
+
+ /* Use a different port base in case VBox already is running
+ * with the same service using ATS_TCP_DEF_BIND_PORT_HOST. */
+ uPort = ATS_TCP_DEF_BIND_PORT_HOST + RTRandU32Ex(0, 4242);
+ }
+
+ if (RT_SUCCESS(rc))
+ {
+ RTTestPrintf(hTest, RTTESTLVL_ALWAYS, "Using port %RU32\n", uPort);
+
+ rc = AudioTestSvcClientCreate(&Client);
+ RTTEST_CHECK_RC_OK(hTest, rc);
+
+ Val.u32 = ATSCONNMODE_CLIENT;
+ rc = AudioTestSvcClientHandleOption(&Client, ATSTCPOPT_CONN_MODE, &Val);
+ RTTEST_CHECK_RC_OK(hTest, rc);
+
+ Val.psz = ATS_TCP_DEF_CONNECT_HOST_ADDR_STR;
+ rc = AudioTestSvcClientHandleOption(&Client, ATSTCPOPT_CONNECT_ADDRESS, &Val);
+ RTTEST_CHECK_RC_OK(hTest, rc);
+
+ Val.u16 = uPort;
+ rc = AudioTestSvcClientHandleOption(&Client, ATSTCPOPT_CONNECT_PORT, &Val);
+ RTTEST_CHECK_RC_OK(hTest, rc);
+
+ rc = AudioTestSvcClientConnect(&Client);
+ RTTEST_CHECK_RC_OK(hTest, rc);
+ }
+ }
+
+ if (RT_SUCCESS(rc))
+ {
+ char szTemp[RTPATH_MAX];
+ rc = RTPathTemp(szTemp, sizeof(szTemp));
+ RTTEST_CHECK_RC_OK(hTest, rc);
+ if (RT_SUCCESS(rc))
+ {
+ char szName[RTPATH_MAX];
+ RTTESTI_CHECK_RC(rc = RTStrCopy(szName, sizeof(szName), szTemp), VINF_SUCCESS);
+ RTTESTI_CHECK_RC(rc = RTPathAppend(szName, sizeof(szName), "tstAudioTestService-XXXXX"), VINF_SUCCESS);
+ if (RT_SUCCESS(rc))
+ {
+ rc = AudioTestSvcClientTestSetDownload(&Client, "ignored", szName);
+ RTTEST_CHECK_RC_OK(hTest, rc);
+
+ /* ignore rc */ RTFileDelete(szName);
+ }
+ }
+ }
+
+ rc = AudioTestSvcClientDisconnect(&Client);
+ RTTEST_CHECK_RC_OK(hTest, rc);
+
+ AudioTestSvcClientDestroy(&Client);
+
+ rc = AudioTestSvcStop(&Srv);
+ RTTEST_CHECK_RC_OK(hTest, rc);
+
+ rc = AudioTestSvcDestroy(&Srv);
+ RTTEST_CHECK_RC_OK(hTest, rc);
+
+ /*
+ * Summary
+ */
+ return RTTestSummaryAndDestroy(hTest);
+}