henrika 9b2b40231d Ensures that RECORD_AUDIO permission is required to start recording. ...

Avoids existing crash and ensures that error message is passed up to Libjingle. Will lead to the following logcat output:

E/libjingle(31404): Error(channel.cc:1514): Failed to SetSend 2 on voice channel

BUG=b/21273153
R=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54459004

Cr-Commit-Position: refs/heads/master@{#9236}

2015-05-20 14:08:44 +00:00

..

audio_coding

Clear ARM NEON flag

2015-05-20 05:20:04 +00:00

audio_conference_mixer

Remove henrike@ from OWNERS

2015-04-01 15:08:49 +00:00

audio_device

Ensures that RECORD_AUDIO permission is required to start recording.

2015-05-20 14:08:44 +00:00

audio_processing

Replace assert() with static_assert() if the condition is evaluatable at

2015-05-20 09:11:02 +00:00

bitrate_controller

Moving the pacer and the pacer thread to ChannelGroup.

2015-03-26 10:11:22 +00:00

desktop_capture

Make the BlockDifference() functions return DiffInfo as their callers expect.

2015-05-18 21:39:07 +00:00

interface

Record H264 NALU type in the h264 header.

2015-04-23 18:14:46 +00:00

media_file

Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro

2015-03-04 13:04:54 +00:00

pacing

Use padding to achieve bitrate probing if the initial key frame has too few packets.

2015-05-05 08:21:32 +00:00

remote_bitrate_estimator

Wire up RTT to send-side GCC and TCP.

2015-05-07 13:04:29 +00:00

rtp_rtcp

Refactor RTCP sender

2015-05-11 08:17:46 +00:00

utility

Remove rtp_rtcp/ dump functionality.

2015-05-20 12:10:56 +00:00

video_capture

Move SetCurrentThreadName to platform_thread.* in rtc_base_approved,

2015-05-18 11:50:31 +00:00

video_coding

Remove VCM debug recordings.

2015-05-20 11:47:26 +00:00

video_processing

Remove VideoProcessing::ColorEnhancement.

2015-05-13 09:27:14 +00:00

video_render

Move IncomingVideoFrames to common_video/.

2015-05-13 11:28:04 +00:00

module_common_types_unittest.cc

Fix issues with incorrect wrap checks when having big buffers and high bitrate.

2013-04-11 17:48:02 +00:00

modules_java_chromium.gyp

Fixes build breakage when building WebRTC in Chromium and having include_tests=1.

2013-07-08 18:14:58 +00:00

modules_java.gyp

Build all java files into jar for each module on Android

2013-07-01 17:52:39 +00:00

modules_tests.isolate

Update isolate files for Android APK tests.

2014-11-13 08:35:05 +00:00

modules_unittests.isolate

Adds C++/JNI/Java unit test for audio device module on Android.

2015-03-09 12:40:43 +00:00

modules.gyp

Adding support for OpenSL ES output in native WebRTC

2015-05-18 14:49:04 +00:00

OWNERS

Add kjellander@webrtc.org as OWNER for *.isolate

2014-06-10 05:42:53 +00:00