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				https://gitlab.freedesktop.org/pipewire/pipewire.git
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	Codecs may need to fragment a single encoder frame across multiple packets that are sent consecutively. Allow codec->encode() to set need_flush=NEED_FLUSH_FRAGMENT, so that sink should immediately call start_encode + encode with NULL input data, to produce the next packet. Previously, other return values than need_flush=1 were unused, so no need to bump codec ABI for this.
		
			
				
	
	
		
			565 lines
		
	
	
	
		
			15 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			565 lines
		
	
	
	
		
			15 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
/* Spa A2DP AAC codec
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 *
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 * Copyright © 2020 Wim Taymans
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 *
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 * Permission is hereby granted, free of charge, to any person obtaining a
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 * copy of this software and associated documentation files (the "Software"),
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 * to deal in the Software without restriction, including without limitation
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 * the rights to use, copy, modify, merge, publish, distribute, sublicense,
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 * and/or sell copies of the Software, and to permit persons to whom the
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 * Software is furnished to do so, subject to the following conditions:
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 *
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 * The above copyright notice and this permission notice (including the next
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 * paragraph) shall be included in all copies or substantial portions of the
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 * Software.
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 *
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 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.  IN NO EVENT SHALL
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 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
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 * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
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 * DEALINGS IN THE SOFTWARE.
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 */
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#include <unistd.h>
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#include <stddef.h>
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#include <errno.h>
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#include <arpa/inet.h>
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#include <spa/param/audio/format.h>
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#include <spa/utils/dict.h>
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#include <fdk-aac/aacenc_lib.h>
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#include "rtp.h"
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#include "a2dp-codecs.h"
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#define DEFAULT_AAC_BITRATE	320000
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#define MIN_AAC_BITRATE		64000
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struct props {
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	int bitratemode;
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};
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struct impl {
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	HANDLE_AACENCODER aacenc;
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	struct rtp_header *header;
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	size_t mtu;
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	int codesize;
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	int max_bitrate;
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	int cur_bitrate;
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	uint32_t rate;
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	uint32_t channels;
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	int samplesize;
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};
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static int codec_fill_caps(const struct a2dp_codec *codec, uint32_t flags,
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		uint8_t caps[A2DP_MAX_CAPS_SIZE])
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{
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	static const a2dp_aac_t a2dp_aac = {
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		.object_type =
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			/* NOTE: AAC Long Term Prediction and AAC Scalable are
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			 *       not supported by the FDK-AAC library. */
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			AAC_OBJECT_TYPE_MPEG2_AAC_LC |
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			AAC_OBJECT_TYPE_MPEG4_AAC_LC,
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		AAC_INIT_FREQUENCY(
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			AAC_SAMPLING_FREQ_8000 |
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			AAC_SAMPLING_FREQ_11025 |
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			AAC_SAMPLING_FREQ_12000 |
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			AAC_SAMPLING_FREQ_16000 |
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			AAC_SAMPLING_FREQ_22050 |
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			AAC_SAMPLING_FREQ_24000 |
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			AAC_SAMPLING_FREQ_32000 |
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			AAC_SAMPLING_FREQ_44100 |
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			AAC_SAMPLING_FREQ_48000 |
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			AAC_SAMPLING_FREQ_64000 |
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			AAC_SAMPLING_FREQ_88200 |
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			AAC_SAMPLING_FREQ_96000)
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		.channels =
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			AAC_CHANNELS_1 |
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			AAC_CHANNELS_2,
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		.vbr = 1,
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		AAC_INIT_BITRATE(DEFAULT_AAC_BITRATE)
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	};
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	memcpy(caps, &a2dp_aac, sizeof(a2dp_aac));
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	return sizeof(a2dp_aac);
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}
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static const struct a2dp_codec_config
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aac_frequencies[] = {
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	{ AAC_SAMPLING_FREQ_48000, 48000, 11 },
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	{ AAC_SAMPLING_FREQ_44100, 44100, 10 },
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	{ AAC_SAMPLING_FREQ_96000, 96000, 9 },
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	{ AAC_SAMPLING_FREQ_88200, 88200, 8 },
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	{ AAC_SAMPLING_FREQ_64000, 64000, 7 },
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	{ AAC_SAMPLING_FREQ_32000, 32000, 6 },
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	{ AAC_SAMPLING_FREQ_24000, 24000, 5 },
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	{ AAC_SAMPLING_FREQ_22050, 22050, 4 },
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	{ AAC_SAMPLING_FREQ_16000, 16000, 3 },
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	{ AAC_SAMPLING_FREQ_12000, 12000, 2 },
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	{ AAC_SAMPLING_FREQ_11025, 11025, 1 },
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	{ AAC_SAMPLING_FREQ_8000,  8000,  0 },
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};
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static const struct a2dp_codec_config
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aac_channel_modes[] = {
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	{ AAC_CHANNELS_2, 2, 1 },
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	{ AAC_CHANNELS_1, 1, 0 },
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};
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static int get_valid_aac_bitrate(a2dp_aac_t *conf)
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{
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	if (AAC_GET_BITRATE(*conf) < MIN_AAC_BITRATE) {
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		/* Unknown (0) or bogus bitrate */
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		return DEFAULT_AAC_BITRATE;
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	} else {
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		return SPA_MIN(AAC_GET_BITRATE(*conf), DEFAULT_AAC_BITRATE);
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	}
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}
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static int codec_select_config(const struct a2dp_codec *codec, uint32_t flags,
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		const void *caps, size_t caps_size,
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		const struct a2dp_codec_audio_info *info,
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		const struct spa_dict *settings, uint8_t config[A2DP_MAX_CAPS_SIZE])
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{
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	a2dp_aac_t conf;
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	int i;
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	if (caps_size < sizeof(conf))
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		return -EINVAL;
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	conf = *(a2dp_aac_t*)caps;
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	if (conf.object_type & AAC_OBJECT_TYPE_MPEG2_AAC_LC)
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		conf.object_type = AAC_OBJECT_TYPE_MPEG2_AAC_LC;
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	else if (conf.object_type & AAC_OBJECT_TYPE_MPEG4_AAC_LC)
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		conf.object_type = AAC_OBJECT_TYPE_MPEG4_AAC_LC;
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	else if (conf.object_type & AAC_OBJECT_TYPE_MPEG4_AAC_LTP)
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		return -ENOTSUP;  /* Not supported by FDK-AAC */
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	else if (conf.object_type & AAC_OBJECT_TYPE_MPEG4_AAC_SCA)
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		return -ENOTSUP;  /* Not supported by FDK-AAC */
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	else
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		return -ENOTSUP;
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	if ((i = a2dp_codec_select_config(aac_frequencies,
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					  SPA_N_ELEMENTS(aac_frequencies),
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					  AAC_GET_FREQUENCY(conf),
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				    	  info ? info->rate : A2DP_CODEC_DEFAULT_RATE
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				    	  )) < 0)
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		return -ENOTSUP;
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	AAC_SET_FREQUENCY(conf, aac_frequencies[i].config);
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	if ((i = a2dp_codec_select_config(aac_channel_modes,
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					  SPA_N_ELEMENTS(aac_channel_modes),
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					  conf.channels,
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				    	  info ? info->channels : A2DP_CODEC_DEFAULT_CHANNELS
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				    	  )) < 0)
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		return -ENOTSUP;
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	conf.channels = aac_channel_modes[i].config;
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	AAC_SET_BITRATE(conf, get_valid_aac_bitrate(&conf));
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	memcpy(config, &conf, sizeof(conf));
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	return sizeof(conf);
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}
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static int codec_enum_config(const struct a2dp_codec *codec,
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		const void *caps, size_t caps_size, uint32_t id, uint32_t idx,
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		struct spa_pod_builder *b, struct spa_pod **param)
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{
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	a2dp_aac_t conf;
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	struct spa_pod_frame f[2];
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	struct spa_pod_choice *choice;
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	uint32_t position[SPA_AUDIO_MAX_CHANNELS];
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	uint32_t i = 0;
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	if (caps_size < sizeof(conf))
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		return -EINVAL;
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	memcpy(&conf, caps, sizeof(conf));
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	if (idx > 0)
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		return 0;
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	spa_pod_builder_push_object(b, &f[0], SPA_TYPE_OBJECT_Format, id);
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	spa_pod_builder_add(b,
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			SPA_FORMAT_mediaType,      SPA_POD_Id(SPA_MEDIA_TYPE_audio),
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			SPA_FORMAT_mediaSubtype,   SPA_POD_Id(SPA_MEDIA_SUBTYPE_raw),
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			SPA_FORMAT_AUDIO_format,   SPA_POD_Id(SPA_AUDIO_FORMAT_S16),
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			0);
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	spa_pod_builder_prop(b, SPA_FORMAT_AUDIO_rate, 0);
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	spa_pod_builder_push_choice(b, &f[1], SPA_CHOICE_None, 0);
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	choice = (struct spa_pod_choice*)spa_pod_builder_frame(b, &f[1]);
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	i = 0;
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	for (size_t j = 0; j < SPA_N_ELEMENTS(aac_frequencies); j++) {
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		if (AAC_GET_FREQUENCY(conf) & aac_frequencies[j].config) {
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			if (i++ == 0)
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				spa_pod_builder_int(b, aac_frequencies[j].value);
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			spa_pod_builder_int(b, aac_frequencies[j].value);
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		}
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	}
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	if (i == 0)
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		return -EINVAL;
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	if (i > 1)
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		choice->body.type = SPA_CHOICE_Enum;
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	spa_pod_builder_pop(b, &f[1]);
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	if (SPA_FLAG_IS_SET(conf.channels, AAC_CHANNELS_1 | AAC_CHANNELS_2)) {
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		spa_pod_builder_add(b,
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				SPA_FORMAT_AUDIO_channels, SPA_POD_CHOICE_RANGE_Int(2, 1, 2),
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				0);
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	} else if (conf.channels & AAC_CHANNELS_1) {
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		position[0] = SPA_AUDIO_CHANNEL_MONO;
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		spa_pod_builder_add(b,
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				SPA_FORMAT_AUDIO_channels, SPA_POD_Int(1),
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				SPA_FORMAT_AUDIO_position, SPA_POD_Array(sizeof(uint32_t),
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					SPA_TYPE_Id, 1, position),
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				0);
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	} else if (conf.channels & AAC_CHANNELS_2) {
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		position[0] = SPA_AUDIO_CHANNEL_FL;
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		position[1] = SPA_AUDIO_CHANNEL_FR;
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		spa_pod_builder_add(b,
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				SPA_FORMAT_AUDIO_channels, SPA_POD_Int(2),
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				SPA_FORMAT_AUDIO_position, SPA_POD_Array(sizeof(uint32_t),
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					SPA_TYPE_Id, 2, position),
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				0);
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	} else
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		return -EINVAL;
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	*param = spa_pod_builder_pop(b, &f[0]);
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	return *param == NULL ? -EIO : 1;
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}
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static int codec_validate_config(const struct a2dp_codec *codec, uint32_t flags,
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			const void *caps, size_t caps_size,
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			struct spa_audio_info *info)
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{
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	a2dp_aac_t conf;
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	size_t j;
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	if (caps == NULL || caps_size < sizeof(conf))
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		return -EINVAL;
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	memcpy(&conf, caps, sizeof(conf));
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	spa_zero(*info);
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	info->media_type = SPA_MEDIA_TYPE_audio;
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	info->media_subtype = SPA_MEDIA_SUBTYPE_raw;
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	info->info.raw.format = SPA_AUDIO_FORMAT_S16;
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	/*
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	 * A2DP v1.3.2, 4.5.2: only one bit shall be set in bitfields.
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	 * However, there is a report (#1342) of device setting multiple
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	 * bits for AAC object type. It's not clear if this was due to
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	 * a BlueZ bug, but we can be lax here and below in codec_init.
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	 */
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	if (!(conf.object_type & (AAC_OBJECT_TYPE_MPEG2_AAC_LC |
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					AAC_OBJECT_TYPE_MPEG4_AAC_LC)))
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		return -EINVAL;
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	for (j = 0; j < SPA_N_ELEMENTS(aac_frequencies); ++j) {
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		if (AAC_GET_FREQUENCY(conf) & aac_frequencies[j].config) {
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			info->info.raw.rate = aac_frequencies[j].value;
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			break;
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		}
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	}
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	if (j == SPA_N_ELEMENTS(aac_frequencies))
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		return -EINVAL;
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	if (conf.channels & AAC_CHANNELS_2) {
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		info->info.raw.channels = 2;
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		info->info.raw.position[0] = SPA_AUDIO_CHANNEL_FL;
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		info->info.raw.position[1] = SPA_AUDIO_CHANNEL_FR;
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	} else if (conf.channels & AAC_CHANNELS_1) {
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		info->info.raw.channels = 1;
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		info->info.raw.position[0] = SPA_AUDIO_CHANNEL_MONO;
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	} else {
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		return -EINVAL;
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	}
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	return 0;
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}
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static void *codec_init_props(const struct a2dp_codec *codec, const struct spa_dict *settings)
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{
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	struct props *p = calloc(1, sizeof(struct props));
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	const char *str;
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	if (p == NULL)
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		return NULL;
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	if (settings == NULL || (str = spa_dict_lookup(settings, "bluez5.a2dp.aac.bitratemode")) == NULL)
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		str = "0";
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	p->bitratemode = SPA_CLAMP(atoi(str), 0, 5);
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	return p;
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}
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static void codec_clear_props(void *props)
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{
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	free(props);
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}
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static void *codec_init(const struct a2dp_codec *codec, uint32_t flags,
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		void *config, size_t config_len, const struct spa_audio_info *info,
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		void *props, size_t mtu)
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{
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	struct impl *this;
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	a2dp_aac_t *conf = config;
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	struct props *p = props;
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	UINT bitratemode;
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	int res;
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	this = calloc(1, sizeof(struct impl));
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	if (this == NULL) {
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		res = -errno;
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		goto error;
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	}
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	this->mtu = mtu;
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	this->rate = info->info.raw.rate;
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	this->channels = info->info.raw.channels;
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	if (info->media_type != SPA_MEDIA_TYPE_audio ||
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	    info->media_subtype != SPA_MEDIA_SUBTYPE_raw ||
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	    info->info.raw.format != SPA_AUDIO_FORMAT_S16) {
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		res = -EINVAL;
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		goto error;
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	}
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	this->samplesize = 2;
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	bitratemode = p ? p->bitratemode : 0;
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	res = aacEncOpen(&this->aacenc, 0, this->channels);
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	if (res != AACENC_OK)
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		goto error;
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 | 
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	if (!(conf->object_type & (AAC_OBJECT_TYPE_MPEG2_AAC_LC |
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					AAC_OBJECT_TYPE_MPEG4_AAC_LC))) {
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		res = -EINVAL;
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		goto error;
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	}
 | 
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 | 
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	res = aacEncoder_SetParam(this->aacenc, AACENC_AOT, AOT_AAC_LC);
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	if (res != AACENC_OK)
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		goto error;
 | 
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 | 
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	res = aacEncoder_SetParam(this->aacenc, AACENC_SAMPLERATE, this->rate);
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	if (res != AACENC_OK)
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		goto error;
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 | 
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	res = aacEncoder_SetParam(this->aacenc, AACENC_CHANNELMODE, this->channels);
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	if (res != AACENC_OK)
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		goto error;
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 | 
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	if (conf->vbr) {
 | 
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		res = aacEncoder_SetParam(this->aacenc, AACENC_BITRATEMODE,
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				bitratemode);
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		if (res != AACENC_OK)
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			goto error;
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	}
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	res = aacEncoder_SetParam(this->aacenc, AACENC_AUDIOMUXVER, 2);
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	if (res != AACENC_OK)
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		goto error;
 | 
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 | 
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	res = aacEncoder_SetParam(this->aacenc, AACENC_SIGNALING_MODE, 1);
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	if (res != AACENC_OK)
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		goto error;
 | 
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 | 
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	// Fragmentation is not implemented yet,
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	// so make sure every encoded AAC frame fits in (mtu - header)
 | 
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	this->max_bitrate = ((this->mtu - sizeof(struct rtp_header)) * 8 * this->rate) / 1024;
 | 
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	this->max_bitrate = SPA_MIN(this->max_bitrate, get_valid_aac_bitrate(conf));
 | 
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	this->cur_bitrate = this->max_bitrate;
 | 
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 | 
						|
	res = aacEncoder_SetParam(this->aacenc, AACENC_BITRATE, this->cur_bitrate);
 | 
						|
	if (res != AACENC_OK)
 | 
						|
		goto error;
 | 
						|
 | 
						|
	res = aacEncoder_SetParam(this->aacenc, AACENC_PEAK_BITRATE, this->max_bitrate);
 | 
						|
	if (res != AACENC_OK)
 | 
						|
		goto error;
 | 
						|
 | 
						|
	res = aacEncoder_SetParam(this->aacenc, AACENC_TRANSMUX, TT_MP4_LATM_MCP1);
 | 
						|
	if (res != AACENC_OK)
 | 
						|
		goto error;
 | 
						|
 | 
						|
	res = aacEncoder_SetParam(this->aacenc, AACENC_HEADER_PERIOD, 1);
 | 
						|
	if (res != AACENC_OK)
 | 
						|
		goto error;
 | 
						|
 | 
						|
	res = aacEncoder_SetParam(this->aacenc, AACENC_AFTERBURNER, 1);
 | 
						|
	if (res != AACENC_OK)
 | 
						|
		goto error;
 | 
						|
 | 
						|
	res = aacEncEncode(this->aacenc, NULL, NULL, NULL, NULL);
 | 
						|
	if (res != AACENC_OK)
 | 
						|
		goto error;
 | 
						|
 | 
						|
	AACENC_InfoStruct enc_info = {};
 | 
						|
	res = aacEncInfo(this->aacenc, &enc_info);
 | 
						|
	if (res != AACENC_OK)
 | 
						|
		goto error;
 | 
						|
 | 
						|
	this->codesize = enc_info.frameLength * this->channels * this->samplesize;
 | 
						|
 | 
						|
	return this;
 | 
						|
 | 
						|
error:
 | 
						|
	if (this->aacenc)
 | 
						|
		aacEncClose(&this->aacenc);
 | 
						|
	free(this);
 | 
						|
	errno = -res;
 | 
						|
	return NULL;
 | 
						|
}
 | 
						|
 | 
						|
static void codec_deinit(void *data)
 | 
						|
{
 | 
						|
	struct impl *this = data;
 | 
						|
	if (this->aacenc)
 | 
						|
		aacEncClose(&this->aacenc);
 | 
						|
	free(this);
 | 
						|
}
 | 
						|
 | 
						|
static int codec_get_block_size(void *data)
 | 
						|
{
 | 
						|
	struct impl *this = data;
 | 
						|
	return this->codesize;
 | 
						|
}
 | 
						|
 | 
						|
static int codec_start_encode (void *data,
 | 
						|
		void *dst, size_t dst_size, uint16_t seqnum, uint32_t timestamp)
 | 
						|
{
 | 
						|
	struct impl *this = data;
 | 
						|
 | 
						|
	this->header = (struct rtp_header *)dst;
 | 
						|
	memset(this->header, 0, sizeof(struct rtp_header));
 | 
						|
 | 
						|
	this->header->v = 2;
 | 
						|
	this->header->pt = 96;
 | 
						|
	this->header->sequence_number = htons(seqnum);
 | 
						|
	this->header->timestamp = htonl(timestamp);
 | 
						|
	this->header->ssrc = htonl(1);
 | 
						|
	return sizeof(struct rtp_header);
 | 
						|
}
 | 
						|
 | 
						|
static int codec_encode(void *data,
 | 
						|
		const void *src, size_t src_size,
 | 
						|
		void *dst, size_t dst_size,
 | 
						|
		size_t *dst_out, int *need_flush)
 | 
						|
{
 | 
						|
	struct impl *this = data;
 | 
						|
	int res;
 | 
						|
 | 
						|
	void *in_bufs[] = {(void *) src};
 | 
						|
	int in_buf_ids[] = {IN_AUDIO_DATA};
 | 
						|
	int in_buf_sizes[] = {src_size};
 | 
						|
	int in_buf_el_sizes[] = {this->samplesize};
 | 
						|
	AACENC_BufDesc in_buf_desc = {
 | 
						|
		.numBufs = 1,
 | 
						|
		.bufs = in_bufs,
 | 
						|
		.bufferIdentifiers = in_buf_ids,
 | 
						|
		.bufSizes = in_buf_sizes,
 | 
						|
		.bufElSizes = in_buf_el_sizes,
 | 
						|
	};
 | 
						|
	AACENC_InArgs in_args = {
 | 
						|
		.numInSamples = src_size / this->samplesize,
 | 
						|
	};
 | 
						|
 | 
						|
	void *out_bufs[] = {dst};
 | 
						|
	int out_buf_ids[] = {OUT_BITSTREAM_DATA};
 | 
						|
	int out_buf_sizes[] = {dst_size};
 | 
						|
	int out_buf_el_sizes[] = {this->samplesize};
 | 
						|
	AACENC_BufDesc out_buf_desc = {
 | 
						|
		.numBufs = 1,
 | 
						|
		.bufs = out_bufs,
 | 
						|
		.bufferIdentifiers = out_buf_ids,
 | 
						|
		.bufSizes = out_buf_sizes,
 | 
						|
		.bufElSizes = out_buf_el_sizes,
 | 
						|
	};
 | 
						|
	AACENC_OutArgs out_args = {};
 | 
						|
 | 
						|
	res = aacEncEncode(this->aacenc, &in_buf_desc, &out_buf_desc, &in_args, &out_args);
 | 
						|
	if (res != AACENC_OK)
 | 
						|
		return -EINVAL;
 | 
						|
 | 
						|
	*dst_out = out_args.numOutBytes;
 | 
						|
	*need_flush = NEED_FLUSH_ALL;
 | 
						|
 | 
						|
	/* RFC6416: It is set to 1 to indicate that the RTP packet contains a complete
 | 
						|
   	 * audioMuxElement or the last fragment of an audioMuxElement */
 | 
						|
	this->header->m = 1;
 | 
						|
 | 
						|
	return out_args.numInSamples * this->samplesize;
 | 
						|
}
 | 
						|
 | 
						|
static int codec_abr_process (void *data, size_t unsent)
 | 
						|
{
 | 
						|
	return -ENOTSUP;
 | 
						|
}
 | 
						|
 | 
						|
static int codec_change_bitrate(struct impl *this, int new_bitrate)
 | 
						|
{
 | 
						|
	int res;
 | 
						|
 | 
						|
	new_bitrate = SPA_MIN(new_bitrate, this->max_bitrate);
 | 
						|
	new_bitrate = SPA_MAX(new_bitrate, 64000);
 | 
						|
 | 
						|
	if (new_bitrate == this->cur_bitrate)
 | 
						|
		return 0;
 | 
						|
 | 
						|
	this->cur_bitrate = new_bitrate;
 | 
						|
 | 
						|
	res = aacEncoder_SetParam(this->aacenc, AACENC_BITRATE, this->cur_bitrate);
 | 
						|
	if (res != AACENC_OK)
 | 
						|
		return -EINVAL;
 | 
						|
 | 
						|
	return 0;
 | 
						|
}
 | 
						|
 | 
						|
static int codec_reduce_bitpool(void *data)
 | 
						|
{
 | 
						|
	struct impl *this = data;
 | 
						|
	return codec_change_bitrate(this, (this->cur_bitrate * 2) / 3);
 | 
						|
}
 | 
						|
 | 
						|
static int codec_increase_bitpool(void *data)
 | 
						|
{
 | 
						|
	struct impl *this = data;
 | 
						|
	return codec_change_bitrate(this, (this->cur_bitrate * 4) / 3);
 | 
						|
}
 | 
						|
 | 
						|
const struct a2dp_codec a2dp_codec_aac = {
 | 
						|
	.id = SPA_BLUETOOTH_AUDIO_CODEC_AAC,
 | 
						|
	.codec_id = A2DP_CODEC_MPEG24,
 | 
						|
	.name = "aac",
 | 
						|
	.description = "AAC",
 | 
						|
	.fill_caps = codec_fill_caps,
 | 
						|
	.select_config = codec_select_config,
 | 
						|
	.enum_config = codec_enum_config,
 | 
						|
	.validate_config = codec_validate_config,
 | 
						|
	.init_props = codec_init_props,
 | 
						|
	.clear_props = codec_clear_props,
 | 
						|
	.init = codec_init,
 | 
						|
	.deinit = codec_deinit,
 | 
						|
	.get_block_size = codec_get_block_size,
 | 
						|
	.start_encode = codec_start_encode,
 | 
						|
	.encode = codec_encode,
 | 
						|
	.abr_process = codec_abr_process,
 | 
						|
	.reduce_bitpool = codec_reduce_bitpool,
 | 
						|
	.increase_bitpool = codec_increase_bitpool,
 | 
						|
};
 | 
						|
 | 
						|
A2DP_CODEC_EXPORT_DEF(
 | 
						|
	"aac",
 | 
						|
	&a2dp_codec_aac
 | 
						|
);
 |