ec4fcd4b2a
- update seamonkey to 2.21 - update firefox-esr to 17.0.9 - enable GSTREAMER by default for html5 with h264/aac/mp3 - WEBRTC is now always built - add PROFILE and TESTS options Security: 7dfed67b-20aa-11e3-b8d8-0025905a4771 In collaboration with: Jan Beich <jbeich@tormail.org>
103 lines
4.3 KiB
Plaintext
103 lines
4.3 KiB
Plaintext
diff --git media/webrtc/trunk/webrtc/modules/audio_device/audio_device_impl.cc media/webrtc/trunk/webrtc/modules/audio_device/audio_device_impl.cc
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index f231b1e..6087696 100644
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--- media/webrtc/trunk/webrtc/modules/audio_device/audio_device_impl.cc
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+++ media/webrtc/trunk/webrtc/modules/audio_device/audio_device_impl.cc
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@@ -16,7 +16,9 @@
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#include <assert.h>
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#include <string.h>
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-#if defined(_WIN32)
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+#if defined(WEBRTC_DUMMY_AUDIO_BUILD)
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+// do not include platform specific headers
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+#elif defined(_WIN32)
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#include "audio_device_utility_win.h"
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#include "audio_device_wave_win.h"
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#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
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@@ -32,14 +34,8 @@
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#include <stdlib.h>
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#include "audio_device_utility_android.h"
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#include "audio_device_jni_android.h"
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-#elif defined(WEBRTC_LINUX)
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+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
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#include "audio_device_utility_linux.h"
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- #if defined(LINUX_ALSA)
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- #include "audio_device_alsa_linux.h"
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- #endif
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- #if defined(LINUX_PULSE)
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- #include "audio_device_pulse_linux.h"
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- #endif
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#elif defined(WEBRTC_IOS)
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#include "audio_device_utility_ios.h"
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#include "audio_device_ios.h"
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@@ -47,6 +43,12 @@
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#include "audio_device_utility_mac.h"
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#include "audio_device_mac.h"
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#endif
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+#if defined(LINUX_ALSA)
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+ #include "audio_device_alsa_linux.h"
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+#endif
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+#if defined(LINUX_PULSE)
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+ #include "audio_device_pulse_linux.h"
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+#endif
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#include "audio_device_dummy.h"
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#include "audio_device_utility_dummy.h"
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#include "critical_section_wrapper.h"
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@@ -161,7 +163,7 @@ WebRtc_Word32 AudioDeviceModuleImpl::Che
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#elif defined(WEBRTC_ANDROID)
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platform = kPlatformAndroid;
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WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "current platform is ANDROID");
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-#elif defined(WEBRTC_LINUX)
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+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
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platform = kPlatformLinux;
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WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "current platform is LINUX");
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#elif defined(WEBRTC_IOS)
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@@ -309,7 +311,7 @@ int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects()
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// Create the *Linux* implementation of the Audio Device
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//
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-#elif defined(WEBRTC_LINUX)
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+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
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if ((audioLayer == kLinuxPulseAudio) || (audioLayer == kPlatformDefaultAudio))
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{
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#if defined(LINUX_PULSE)
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@@ -355,7 +357,7 @@ int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects()
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//
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ptrAudioDeviceUtility = new AudioDeviceUtilityLinux(Id());
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}
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-#endif // #if defined(WEBRTC_LINUX)
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+#endif // #if defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
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// Create the *iPhone* implementation of the Audio Device
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//
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diff --git media/webrtc/trunk/webrtc/modules/audio_device/test/audio_device_test_api.cc media/webrtc/trunk/webrtc/modules/audio_device/test/audio_device_test_api.cc
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index defd7f8..906c4a2 100644
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--- media/webrtc/trunk/webrtc/modules/audio_device/test/audio_device_test_api.cc
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+++ media/webrtc/trunk/webrtc/modules/audio_device/test/audio_device_test_api.cc
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@@ -197,7 +197,7 @@ class AudioDeviceAPITest: public testing::Test {
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// Create default implementation instance
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EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
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kId, AudioDeviceModule::kPlatformDefaultAudio)) != NULL);
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-#elif defined(WEBRTC_LINUX)
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+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
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EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
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kId, AudioDeviceModule::kWindowsWaveAudio)) == NULL);
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EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
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@@ -1690,7 +1690,7 @@ TEST_F(AudioDeviceAPITest, CPULoad) {
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// TODO(kjellander): Fix flakiness causing failures on Windows.
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// TODO(phoglund): Fix flakiness causing failures on Linux.
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-#if !defined(_WIN32) && !defined(WEBRTC_LINUX)
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+#if !defined(_WIN32) && !defined(WEBRTC_LINUX) && !defined(WEBRTC_BSD)
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TEST_F(AudioDeviceAPITest, StartAndStopRawOutputFileRecording) {
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// NOTE: this API is better tested in a functional test
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CheckInitialPlayoutStates();
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@@ -1759,7 +1759,7 @@ TEST_F(AudioDeviceAPITest, StartAndStopRawInputFileRecording) {
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// - size of raw_input_not_recording.pcm shall be 0
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// - size of raw_input_not_recording.pcm shall be > 0
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}
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-#endif // !WIN32 && !WEBRTC_LINUX
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+#endif // !WIN32 && !WEBRTC_LINUX && !defined(WEBRTC_BSD)
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TEST_F(AudioDeviceAPITest, RecordingSampleRate) {
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uint32_t sampleRate(0);
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