mirror of
https://github.com/archlinuxarm/PKGBUILDs.git
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97 lines
4.1 KiB
Diff
97 lines
4.1 KiB
Diff
From ed5409740e3f6c13895a29b770d46d659973863e Mon Sep 17 00:00:00 2001
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From: "dalecurtis@google.com"
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<dalecurtis@google.com@0039d316-1c4b-4281-b951-d872f2087c98>
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Date: Fri, 24 May 2013 23:54:45 +0000
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Subject: [PATCH] Allow sample rate pass through on Linux.
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Requiring the native output sample rate to avoid glitching appears
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to have just been an issue with a poor tlength choice. xians and
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I resolved this prior to turning PulseAudio on by default, but left
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the native sample rate requirement in at the time.
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Since "native sample rate" can change on the fly with PulseAudio
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our approach of specifying FIX_RATE was incorrect. Allowing Pulse
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to handle resampling lets us remove the FIX_RATE flag. It also
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improves CPU usage in cases where Pulse can configure the output
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device to match the requested sample rate.
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BUG=229918
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TEST=extensive manual checks for glitching.
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R=xians@chromium.org
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Review URL: https://codereview.chromium.org/15957002
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git-svn-id: svn://svn.chromium.org/chrome/trunk/src@202233 0039d316-1c4b-4281-b951-d872f2087c98
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---
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content/renderer/media/audio_renderer_mixer_manager.cc | 6 +++---
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media/audio/pulse/audio_manager_pulse.cc | 6 +++++-
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media/audio/pulse/pulse_util.cc | 6 +++---
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3 files changed, 11 insertions(+), 7 deletions(-)
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diff --git a/content/renderer/media/audio_renderer_mixer_manager.cc b/content/renderer/media/audio_renderer_mixer_manager.cc
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index 7db67d1..0044d25 100644
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--- a/content/renderer/media/audio_renderer_mixer_manager.cc
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+++ b/content/renderer/media/audio_renderer_mixer_manager.cc
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@@ -52,9 +52,9 @@ media::AudioRendererMixer* AudioRendererMixerManager::GetMixer(
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return it->second.mixer;
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}
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- // On ChromeOS we can rely on the playback device to handle resampling, so
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- // don't waste cycles on it here.
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-#if defined(OS_CHROMEOS)
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+ // On ChromeOS and Linux we can rely on the playback device to handle
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+ // resampling, so don't waste cycles on it here.
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+#if defined(OS_CHROMEOS) || defined(OS_LINUX)
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int sample_rate = params.sample_rate();
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#else
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int sample_rate = hardware_config_->GetOutputSampleRate();
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diff --git a/media/audio/pulse/audio_manager_pulse.cc b/media/audio/pulse/audio_manager_pulse.cc
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index c127d17..fed919a 100644
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--- a/media/audio/pulse/audio_manager_pulse.cc
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+++ b/media/audio/pulse/audio_manager_pulse.cc
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@@ -141,11 +141,15 @@ AudioParameters AudioManagerPulse::GetPreferredOutputStreamParameters(
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int buffer_size = kDefaultOutputBufferSize;
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int bits_per_sample = 16;
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int input_channels = 0;
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+ int sample_rate;
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if (input_params.IsValid()) {
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bits_per_sample = input_params.bits_per_sample();
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channel_layout = input_params.channel_layout();
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input_channels = input_params.input_channels();
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buffer_size = std::min(buffer_size, input_params.frames_per_buffer());
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+ sample_rate = input_params.sample_rate();
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+ } else {
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+ sample_rate = GetNativeSampleRate();
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}
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int user_buffer_size = GetUserBufferSize();
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@@ -154,7 +158,7 @@ AudioParameters AudioManagerPulse::GetPreferredOutputStreamParameters(
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return AudioParameters(
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AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, input_channels,
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- GetNativeSampleRate(), bits_per_sample, buffer_size);
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+ sample_rate, bits_per_sample, buffer_size);
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}
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AudioOutputStream* AudioManagerPulse::MakeOutputStream(
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diff --git a/media/audio/pulse/pulse_util.cc b/media/audio/pulse/pulse_util.cc
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index 0a4734b..6a4dad1 100644
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--- a/media/audio/pulse/pulse_util.cc
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+++ b/media/audio/pulse/pulse_util.cc
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@@ -289,9 +289,9 @@ bool CreateOutputStream(pa_threaded_mainloop** mainloop,
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pa_stream_connect_playback(
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*stream, NULL, &pa_buffer_attributes,
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static_cast<pa_stream_flags_t>(
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- PA_STREAM_FIX_RATE | PA_STREAM_INTERPOLATE_TIMING |
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- PA_STREAM_ADJUST_LATENCY | PA_STREAM_AUTO_TIMING_UPDATE |
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- PA_STREAM_NOT_MONOTONIC | PA_STREAM_START_CORKED),
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+ PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_ADJUST_LATENCY |
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+ PA_STREAM_AUTO_TIMING_UPDATE | PA_STREAM_NOT_MONOTONIC |
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+ PA_STREAM_START_CORKED),
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NULL, NULL) == 0,
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"pa_stream_connect_playback FAILED ");
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--
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1.8.2.2
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