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chromium: fix webrtc interaction with pulseaudio

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The webrtc code suffered from a race condition when used
with Pulseaudio. This lead to audio input breaking every
couple of minutes during a webrtc session.
This commit is contained in:
Peter Simons 2020-03-23 15:08:45 +01:00
parent 2a5ef17a63
commit 81b18c3711
2 changed files with 63 additions and 0 deletions
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2
pkgs/applications/networking/browsers/chromium/common.nix
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@ -151,6 +151,8 @@ let
] ++ optionals (useVaapi) [
# source: https://aur.archlinux.org/cgit/aur.git/tree/vaapi-fix.patch?h=chromium-vaapi
./patches/vaapi-fix.patch
# fix race condition in the interaction with pulseaudio
./patches/webrtc-pulse.patch
];
postPatch = ''

61
pkgs/applications/networking/browsers/chromium/patches/webrtc-pulse.patch Normal file
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@ -0,0 +1,61 @@
From 704dc99bd05a94eb61202e6127df94ddfd571e85 Mon Sep 17 00:00:00 2001
From: Dale Curtis <dalecurtis@chromium.org>
Date: Mon, 02 Mar 2020 22:12:22 +0000
Subject: [PATCH] Hold PulseAudio mainloop lock while querying input device info.
a22cc23955cb3d58b7525c5103314226b3ce0137 moved this section out of
UpdateNativeAudioHardwareInfo(), but forgot to bring the lock along.
R=guidou
Bug: 1043040
Change-Id: I5b17a2cf0ad55d61c0811db1dae7045af4a91370
Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/2083814
Commit-Queue: Dale Curtis <dalecurtis@chromium.org>
Commit-Queue: Guido Urdaneta <guidou@chromium.org>
Reviewed-by: Guido Urdaneta <guidou@chromium.org>
Auto-Submit: Dale Curtis <dalecurtis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#746115}
---
diff --git a/media/audio/pulse/audio_manager_pulse.cc b/media/audio/pulse/audio_manager_pulse.cc
index 90e9317..829846f 100644
--- a/media/audio/pulse/audio_manager_pulse.cc
+++ b/media/audio/pulse/audio_manager_pulse.cc
@@ -104,22 +104,27 @@
AudioParameters AudioManagerPulse::GetInputStreamParameters(
const std::string& device_id) {
- int user_buffer_size = GetUserBufferSize();
- int buffer_size =
- user_buffer_size ? user_buffer_size : kDefaultInputBufferSize;
-
UpdateNativeAudioHardwareInfo();
- auto* operation = pa_context_get_source_info_by_name(
- input_context_, default_source_name_.c_str(), DefaultSourceInfoCallback,
- this);
- WaitForOperationCompletion(input_mainloop_, operation, input_context_);
+
+ {
+ AutoPulseLock auto_lock(input_mainloop_);
+ auto* operation = pa_context_get_source_info_by_name(
+ input_context_, default_source_name_.c_str(), DefaultSourceInfoCallback,
+ this);
+ WaitForOperationCompletion(input_mainloop_, operation, input_context_);
+ }
// We don't want to accidentally open a monitor device, so return invalid
- // parameters for those.
+ // parameters for those. Note: The value of |default_source_is_monitor_|
+ // depends on the the call to pa_context_get_source_info_by_name() above.
if (device_id == AudioDeviceDescription::kDefaultDeviceId &&
default_source_is_monitor_) {
return AudioParameters();
}
+
+ const int user_buffer_size = GetUserBufferSize();
+ const int buffer_size =
+ user_buffer_size ? user_buffer_size : kDefaultInputBufferSize;
return AudioParameters(AudioParameters::AUDIO_PCM_LOW_LATENCY,
CHANNEL_LAYOUT_STEREO,
native_input_sample_rate_ ? native_input_sample_rate_