diff options
author | wolfbeast <mcwerewolf@gmail.com> | 2018-05-12 14:32:03 +0200 |
---|---|---|
committer | wolfbeast <mcwerewolf@gmail.com> | 2018-05-12 14:32:03 +0200 |
commit | b7d9dad58e5a3f87a6c767412941700bc8010044 (patch) | |
tree | f5e99b3029cf54409ae5951e9e91cca3a54addc7 /dom/media | |
parent | 73cdd6117df7c17b76aad93952cf574c494351aa (diff) | |
download | UXP-b7d9dad58e5a3f87a6c767412941700bc8010044.tar UXP-b7d9dad58e5a3f87a6c767412941700bc8010044.tar.gz UXP-b7d9dad58e5a3f87a6c767412941700bc8010044.tar.lz UXP-b7d9dad58e5a3f87a6c767412941700bc8010044.tar.xz UXP-b7d9dad58e5a3f87a6c767412941700bc8010044.zip |
Remove MOZ_B2G leftovers and some dead B2G-only components.
Diffstat (limited to 'dom/media')
-rw-r--r-- | dom/media/AudioStream.cpp | 4 | ||||
-rw-r--r-- | dom/media/CubebUtils.cpp | 27 | ||||
-rw-r--r-- | dom/media/CubebUtils.h | 3 | ||||
-rw-r--r-- | dom/media/GraphDriver.cpp | 4 | ||||
-rw-r--r-- | dom/media/MediaManager.cpp | 8 | ||||
-rw-r--r-- | dom/media/PeerConnection.js | 1 | ||||
-rw-r--r-- | dom/media/webrtc/MediaEngine.h | 5 | ||||
-rw-r--r-- | dom/media/webrtc/MediaEngineWebRTCAudio.cpp | 4 |
8 files changed, 1 insertions, 55 deletions
diff --git a/dom/media/AudioStream.cpp b/dom/media/AudioStream.cpp index 82ae3ee86..54cf7b965 100644 --- a/dom/media/AudioStream.cpp +++ b/dom/media/AudioStream.cpp @@ -333,11 +333,7 @@ AudioStream::Init(uint32_t aNumChannels, uint32_t aRate, params.rate = aRate; params.channels = mOutChannels; #if defined(__ANDROID__) -#if defined(MOZ_B2G) - params.stream_type = CubebUtils::ConvertChannelToCubebType(aAudioChannel); -#else params.stream_type = CUBEB_STREAM_TYPE_MUSIC; -#endif if (params.stream_type == CUBEB_STREAM_TYPE_MAX) { return NS_ERROR_INVALID_ARG; diff --git a/dom/media/CubebUtils.cpp b/dom/media/CubebUtils.cpp index d1b4bae99..fe94264ee 100644 --- a/dom/media/CubebUtils.cpp +++ b/dom/media/CubebUtils.cpp @@ -342,33 +342,6 @@ uint32_t MaxNumberOfChannels() return 0; } -#if defined(__ANDROID__) && defined(MOZ_B2G) -cubeb_stream_type ConvertChannelToCubebType(dom::AudioChannel aChannel) -{ - switch(aChannel) { - case dom::AudioChannel::Normal: - /* FALLTHROUGH */ - case dom::AudioChannel::Content: - return CUBEB_STREAM_TYPE_MUSIC; - case dom::AudioChannel::Notification: - return CUBEB_STREAM_TYPE_NOTIFICATION; - case dom::AudioChannel::Alarm: - return CUBEB_STREAM_TYPE_ALARM; - case dom::AudioChannel::Telephony: - return CUBEB_STREAM_TYPE_VOICE_CALL; - case dom::AudioChannel::Ringer: - return CUBEB_STREAM_TYPE_RING; - case dom::AudioChannel::System: - return CUBEB_STREAM_TYPE_SYSTEM; - case dom::AudioChannel::Publicnotification: - return CUBEB_STREAM_TYPE_SYSTEM_ENFORCED; - default: - NS_ERROR("The value of AudioChannel is invalid"); - return CUBEB_STREAM_TYPE_MAX; - } -} -#endif - void GetCurrentBackend(nsAString& aBackend) { cubeb* cubebContext = GetCubebContext(); diff --git a/dom/media/CubebUtils.h b/dom/media/CubebUtils.h index 171c244b7..fa5fc2294 100644 --- a/dom/media/CubebUtils.h +++ b/dom/media/CubebUtils.h @@ -40,9 +40,6 @@ void ReportCubebBackendUsed(); uint32_t GetCubebPlaybackLatencyInMilliseconds(); Maybe<uint32_t> GetCubebMSGLatencyInFrames(); bool CubebLatencyPrefSet(); -#if defined(__ANDROID__) && defined(MOZ_B2G) -cubeb_stream_type ConvertChannelToCubebType(dom::AudioChannel aChannel); -#endif void GetCurrentBackend(nsAString& aBackend); } // namespace CubebUtils diff --git a/dom/media/GraphDriver.cpp b/dom/media/GraphDriver.cpp index 40e3b72cf..cae15eb8c 100644 --- a/dom/media/GraphDriver.cpp +++ b/dom/media/GraphDriver.cpp @@ -640,11 +640,7 @@ AudioCallbackDriver::Init() mSampleRate = output.rate = CubebUtils::PreferredSampleRate(); #if defined(__ANDROID__) -#if defined(MOZ_B2G) - output.stream_type = CubebUtils::ConvertChannelToCubebType(mAudioChannel); -#else output.stream_type = CUBEB_STREAM_TYPE_MUSIC; -#endif if (output.stream_type == CUBEB_STREAM_TYPE_MAX) { NS_WARNING("Bad stream type"); return; diff --git a/dom/media/MediaManager.cpp b/dom/media/MediaManager.cpp index 97a6855d9..44f330e99 100644 --- a/dom/media/MediaManager.cpp +++ b/dom/media/MediaManager.cpp @@ -73,10 +73,6 @@ #include "browser_logging/WebRtcLog.h" #endif -#ifdef MOZ_B2G -#include "MediaPermissionGonk.h" -#endif - #if defined (XP_WIN) #include "mozilla/WindowsVersion.h" #include <winsock2.h> @@ -1819,10 +1815,6 @@ MediaManager::Get() { __LINE__, NS_LITERAL_STRING("Media shutdown")); MOZ_RELEASE_ASSERT(NS_SUCCEEDED(rv)); -#ifdef MOZ_B2G - // Init MediaPermissionManager before sending out any permission requests. - (void) MediaPermissionManager::GetInstance(); -#endif //MOZ_B2G } return sSingleton; } diff --git a/dom/media/PeerConnection.js b/dom/media/PeerConnection.js index 98b8debbe..0c3021799 100644 --- a/dom/media/PeerConnection.js +++ b/dom/media/PeerConnection.js @@ -791,7 +791,6 @@ RTCPeerConnection.prototype = { return this._havePermission; } if (this._isChrome || - AppConstants.MOZ_B2G || Services.prefs.getBoolPref("media.navigator.permission.disabled")) { return this._havePermission = Promise.resolve(); } diff --git a/dom/media/webrtc/MediaEngine.h b/dom/media/webrtc/MediaEngine.h index ff2a6e25a..6a6988544 100644 --- a/dom/media/webrtc/MediaEngine.h +++ b/dom/media/webrtc/MediaEngine.h @@ -54,11 +54,8 @@ public: static const int DEFAULT_169_VIDEO_WIDTH = 1280; static const int DEFAULT_169_VIDEO_HEIGHT = 720; -#ifndef MOZ_B2G static const int DEFAULT_SAMPLE_RATE = 32000; -#else - static const int DEFAULT_SAMPLE_RATE = 16000; -#endif + // This allows using whatever rate the graph is using for the // MediaStreamTrack. This is useful for microphone data, we know it's already // at the correct rate for insertion in the MSG. diff --git a/dom/media/webrtc/MediaEngineWebRTCAudio.cpp b/dom/media/webrtc/MediaEngineWebRTCAudio.cpp index 0b8796aa8..1e2e13d01 100644 --- a/dom/media/webrtc/MediaEngineWebRTCAudio.cpp +++ b/dom/media/webrtc/MediaEngineWebRTCAudio.cpp @@ -741,9 +741,6 @@ MediaEngineWebRTCMicrophoneSource::AllocChannel() // Check for availability. if (!mAudioInput->SetRecordingDevice(mCapIndex)) { -#ifndef MOZ_B2G - // Because of the permission mechanism of B2G, we need to skip the status - // check here. bool avail = false; mAudioInput->GetRecordingDeviceStatus(avail); if (!avail) { @@ -752,7 +749,6 @@ MediaEngineWebRTCMicrophoneSource::AllocChannel() } return false; } -#endif // MOZ_B2G // Set "codec" to PCM, 32kHz on 1 channel ScopedCustomReleasePtr<webrtc::VoECodec> ptrVoECodec(webrtc::VoECodec::GetInterface(mVoiceEngine)); |