mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-24 01:00:37 +00:00
8ca40fa86f
Fixes stuttering audio when iOS AU is resampling. To make AU resample, one has to request a rate that differs from AVAudioSession's sampleRate. The resampling itself is not the culprit, but rather our API misuse. AudioUnitRender modifies the mDataByteSize members with the actual read bytes count. Therefore, they must be reinitialized before each AudioUnitRender. (The buffers themselves can be preallocated.) The "stutter" was caused by one AudioUnitRender making the buffer too small for other AudioUnitRender invocations, making them fail with -50 (paramErr). By way of luck, when AU didn't resample, all AudioUnitRender invocations read the same number of bytes. (This patch addresses some non-interleaved audio concerns, but at this moment the elements do not support non-interleaved audio and non-interleaved is untested.) https://bugzilla.gnome.org/show_bug.cgi?id=744922
328 lines
9 KiB
C
328 lines
9 KiB
C
/*
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* GStreamer
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* Copyright (C) 2012-2013 Fluendo S.A. <support@fluendo.com>
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* Authors: Josep Torra Vallès <josep@fluendo.com>
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* Andoni Morales Alastruey <amorales@fluendo.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*
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*/
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#include "gstosxcoreaudio.h"
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#include "gstosxcoreaudiocommon.h"
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GST_DEBUG_CATEGORY_STATIC (osx_audio_debug);
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#define GST_CAT_DEFAULT osx_audio_debug
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G_DEFINE_TYPE (GstCoreAudio, gst_core_audio, G_TYPE_OBJECT);
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#ifdef HAVE_IOS
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#include "gstosxcoreaudioremoteio.c"
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#else
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#include "gstosxcoreaudiohal.c"
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#endif
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static void
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gst_core_audio_class_init (GstCoreAudioClass * klass)
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{
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}
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static void
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gst_core_audio_init (GstCoreAudio * core_audio)
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{
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core_audio->is_passthrough = FALSE;
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core_audio->device_id = kAudioDeviceUnknown;
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core_audio->is_src = FALSE;
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core_audio->audiounit = NULL;
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#ifndef HAVE_IOS
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core_audio->hog_pid = -1;
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core_audio->disabled_mixing = FALSE;
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#endif
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}
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/**************************
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* Public API *
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*************************/
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GstCoreAudio *
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gst_core_audio_new (GstObject * osxbuf)
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{
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GstCoreAudio *core_audio;
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core_audio = g_object_new (GST_TYPE_CORE_AUDIO, NULL);
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core_audio->osxbuf = osxbuf;
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return core_audio;
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}
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gboolean
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gst_core_audio_close (GstCoreAudio * core_audio)
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{
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AudioComponentInstanceDispose (core_audio->audiounit);
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core_audio->audiounit = NULL;
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return TRUE;
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}
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gboolean
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gst_core_audio_open (GstCoreAudio * core_audio)
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{
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return gst_core_audio_open_impl (core_audio);
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}
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gboolean
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gst_core_audio_start_processing (GstCoreAudio * core_audio)
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{
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return gst_core_audio_start_processing_impl (core_audio);
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}
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gboolean
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gst_core_audio_pause_processing (GstCoreAudio * core_audio)
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{
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return gst_core_audio_pause_processing_impl (core_audio);
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}
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gboolean
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gst_core_audio_stop_processing (GstCoreAudio * core_audio)
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{
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return gst_core_audio_stop_processing_impl (core_audio);
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}
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gboolean
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gst_core_audio_get_samples_and_latency (GstCoreAudio * core_audio,
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gdouble rate, guint * samples, gdouble * latency)
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{
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return gst_core_audio_get_samples_and_latency_impl (core_audio, rate,
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samples, latency);
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}
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gboolean
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gst_core_audio_initialize (GstCoreAudio * core_audio,
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AudioStreamBasicDescription format, GstCaps * caps, gboolean is_passthrough)
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{
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guint32 frame_size;
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OSStatus status;
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GST_DEBUG_OBJECT (core_audio,
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"Initializing: passthrough:%d caps:%" GST_PTR_FORMAT, is_passthrough,
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caps);
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if (!gst_core_audio_initialize_impl (core_audio, format, caps,
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is_passthrough, &frame_size)) {
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goto error;
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}
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if (core_audio->is_src) {
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/* create AudioBufferList needed for recording */
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core_audio->recBufferSize = frame_size * format.mBytesPerFrame;
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core_audio->recBufferList =
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buffer_list_alloc (format.mChannelsPerFrame, core_audio->recBufferSize,
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/* Currently always TRUE (i.e. interleaved) */
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!(format.mFormatFlags & kAudioFormatFlagIsNonInterleaved));
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}
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/* Initialize the AudioUnit */
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status = AudioUnitInitialize (core_audio->audiounit);
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if (status) {
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GST_ERROR_OBJECT (core_audio, "Failed to initialise AudioUnit: %d",
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(int) status);
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goto error;
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}
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return TRUE;
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error:
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buffer_list_free (core_audio->recBufferList);
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core_audio->recBufferList = NULL;
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return FALSE;
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}
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void
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gst_core_audio_unitialize (GstCoreAudio * core_audio)
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{
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AudioUnitUninitialize (core_audio->audiounit);
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buffer_list_free (core_audio->recBufferList);
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core_audio->recBufferList = NULL;
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}
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void
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gst_core_audio_set_volume (GstCoreAudio * core_audio, gfloat volume)
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{
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AudioUnitSetParameter (core_audio->audiounit, kHALOutputParam_Volume,
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kAudioUnitScope_Global, 0, (float) volume, 0);
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}
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gboolean
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gst_core_audio_select_device (GstCoreAudio * core_audio)
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{
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return gst_core_audio_select_device_impl (core_audio);
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}
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void
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gst_core_audio_init_debug (void)
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{
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GST_DEBUG_CATEGORY_INIT (osx_audio_debug, "osxaudio", 0,
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"OSX Audio Elements");
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}
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gboolean
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gst_core_audio_audio_device_is_spdif_avail (AudioDeviceID device_id)
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{
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return gst_core_audio_audio_device_is_spdif_avail_impl (device_id);
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}
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gboolean
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gst_core_audio_parse_channel_layout (AudioChannelLayout * layout,
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gint channels, guint64 * channel_mask, GstAudioChannelPosition * pos)
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{
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gint i;
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gboolean ret = TRUE;
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g_return_val_if_fail (channels <= GST_OSX_AUDIO_MAX_CHANNEL, FALSE);
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switch (channels) {
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case 0:
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pos[0] = GST_AUDIO_CHANNEL_POSITION_NONE;
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break;
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case 1:
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pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
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break;
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case 2:
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pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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*channel_mask |= GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_LEFT);
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*channel_mask |= GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_RIGHT);
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break;
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default:
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for (i = 0; i < channels; i++) {
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switch (layout->mChannelDescriptions[i].mChannelLabel) {
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case kAudioChannelLabel_Left:
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pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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break;
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case kAudioChannelLabel_Right:
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pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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break;
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case kAudioChannelLabel_Center:
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pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
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break;
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case kAudioChannelLabel_LFEScreen:
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pos[i] = GST_AUDIO_CHANNEL_POSITION_LFE1;
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break;
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case kAudioChannelLabel_LeftSurround:
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pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
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break;
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case kAudioChannelLabel_RightSurround:
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pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
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break;
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case kAudioChannelLabel_RearSurroundLeft:
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pos[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT;
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break;
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case kAudioChannelLabel_RearSurroundRight:
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pos[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT;
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break;
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case kAudioChannelLabel_CenterSurround:
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pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
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break;
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default:
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GST_WARNING ("unrecognized channel: %d",
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(int) layout->mChannelDescriptions[i].mChannelLabel);
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*channel_mask = 0;
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ret = FALSE;
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break;
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}
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}
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}
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return ret;
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}
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GstCaps *
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gst_core_audio_asbd_to_caps (AudioStreamBasicDescription * asbd,
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AudioChannelLayout * layout)
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{
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GstAudioInfo info;
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GstAudioFormat format = GST_AUDIO_FORMAT_UNKNOWN;
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GstAudioChannelPosition pos[64] = { GST_AUDIO_CHANNEL_POSITION_INVALID, };
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gint rate, channels, bps, endianness;
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guint64 channel_mask;
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gboolean sign, interleaved;
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if (asbd->mFormatID != kAudioFormatLinearPCM) {
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GST_WARNING ("Only linear PCM is supported");
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goto error;
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}
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if (!(asbd->mFormatFlags & kAudioFormatFlagIsPacked)) {
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GST_WARNING ("Only packed formats supported");
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goto error;
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}
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if (asbd->mFormatFlags & kLinearPCMFormatFlagsSampleFractionMask) {
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GST_WARNING ("Fixed point audio is unsupported");
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goto error;
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}
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rate = asbd->mSampleRate;
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if (rate == kAudioStreamAnyRate)
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rate = GST_AUDIO_DEF_RATE;
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channels = asbd->mChannelsPerFrame;
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if (channels == 0) {
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/* The documentation says this should not happen! */
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channels = 1;
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}
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bps = asbd->mBitsPerChannel;
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endianness = asbd->mFormatFlags & kAudioFormatFlagIsBigEndian ?
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G_BIG_ENDIAN : G_LITTLE_ENDIAN;
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sign = asbd->mFormatID & kAudioFormatFlagIsSignedInteger ? TRUE : FALSE;
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interleaved = asbd->mFormatFlags & kAudioFormatFlagIsNonInterleaved ?
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TRUE : FALSE;
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if (asbd->mFormatFlags & kAudioFormatFlagIsFloat) {
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if (bps == 32) {
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if (endianness == G_LITTLE_ENDIAN)
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format = GST_AUDIO_FORMAT_F32LE;
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else
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format = GST_AUDIO_FORMAT_F32BE;
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} else if (bps == 64) {
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if (endianness == G_LITTLE_ENDIAN)
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format = GST_AUDIO_FORMAT_F64LE;
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else
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format = GST_AUDIO_FORMAT_F64BE;
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}
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} else {
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format = gst_audio_format_build_integer (sign, endianness, bps, bps);
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}
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if (format == GST_AUDIO_FORMAT_UNKNOWN) {
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GST_WARNING ("Unsupported sample format");
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goto error;
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}
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if (!gst_core_audio_parse_channel_layout (layout, channels, &channel_mask,
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pos)) {
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GST_WARNING ("Failed to parse channel layout");
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goto error;
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}
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gst_audio_info_set_format (&info, format, rate, channels, pos);
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return gst_audio_info_to_caps (&info);
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error:
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return NULL;
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}
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