raspi: fix audio decoding with newer libavcodec

Current libavcodec does no longer always return S16 sample format, but
the conversion needs to be done explicitly. Use libswscale for that.
Not tested on raspbian, only on yocto/ffmpeg.

configure.ac

@@ -30,7 +30,6 @@ if test x$BOXTYPE = xgeneric; then
 		PKG_CHECK_MODULES([AVCODEC], [libavcodec >= 54.28.0])
 		# don't know which version is exactly needed here...
 		PKG_CHECK_MODULES([SWSCALE], [libswscale])
-		PKG_CHECK_MODULES([SWRESAMPLE], [libswresample])
 	else
 		# openmaxil are the broadcom userspace libs
 		# my yocto (openembedded) layer has an openmaxil package built from userland git.
@@ -46,6 +45,7 @@ if test x$BOXTYPE = xgeneric; then
 		PKG_CHECK_MODULES([AVFORMAT], [libavformat])
 		PKG_CHECK_MODULES([AVCODEC], [libavcodec])
 	fi
+	PKG_CHECK_MODULES([SWRESAMPLE], [libswresample])
 	# don't know which version is exactly needed here...
 	PKG_CHECK_MODULES([AVUTIL], [libavutil])
 fi

raspi/Makefile.am

@@ -13,6 +13,7 @@ AM_LDFLAGS = \
 	@AVFORMAT_LIBS@ \
 	@AVUTIL_LIBS@ \
 	@AVCODEC_LIBS@ \
+	@SWRESAMPLE_LIBS@ \
 	@OMX_LIBS@ \
 	-lOpenThreads
 

raspi/avdec.cpp

@@ -33,6 +33,7 @@ extern "C" {
 #include <libavutil/opt.h>
 #include <libavutil/samplefmt.h>
 #include <libavutil/mathematics.h>
+#include <libswresample/swresample.h>
 //#include <ao/ao.h>
 #include "codec.h"
 #include "avcodec_omx.h"
@@ -329,6 +330,13 @@ void aDec::run()
 	AVCodecContext *c = NULL;
 	AVInputFormat *inp;
 	AVFrame *frame;
+	/* resample / sample format conversion */
+	SwrContext *swr = NULL;
+	uint8_t *obuf = NULL;
+	int obuf_sz = 0; /* in samples */
+	int obuf_sz_max = 0;
+	int o_ch = 2, o_sr = 48000; /* output channels and sample rate */
+
 	uint8_t *inbuf;
 	AVPacket avpkt;
 	char tmp[128] = "unknown";
@@ -375,6 +383,10 @@ void aDec::run()
 		lt_info("aDec: Codec for codec_id 0x%08x not found\n", c->codec_id);
 		goto out;
 	}
+	/* now using libswresample anyway
+	if (c->sample_fmt == AV_SAMPLE_FMT_S16P)
+		c->request_sample_fmt = AV_SAMPLE_FMT_S16;
+	*/
 	if (avcodec_open2(c, codec, NULL) < 0) {
 		lt_info("aDec: avcodec_open2() failed\n");
 		goto out;
@@ -387,6 +399,15 @@ void aDec::run()
 	/* output sample rate, channels, layout could be set here if necessary */
 	avcodec_string(tmp, sizeof(tmp), c, 0);
 	lt_info("aDec: decoding %s\n", tmp);
+	swr = swr_alloc_set_opts(swr,
+				AV_CH_LAYOUT_STEREO, AV_SAMPLE_FMT_S16, o_sr,		/* output */
+				c->channel_layout, c->sample_fmt, c->sample_rate,	/* input */
+				0, NULL);
+	if (!swr) {
+		lt_info("could not alloc resample context\n");
+		goto out3;
+	}
+	swr_init(swr);
 	while (dec_running) {
 		int gotframe = 0;
 		if (av_read_frame(avfc, &avpkt) < 0) {
@@ -398,19 +419,36 @@ void aDec::run()
 			av_free_packet(&avpkt);
 			continue;
 		}
+
 		int len = avcodec_decode_audio4(c, frame, &gotframe, &avpkt);
 		if (gotframe && dec_running) {
-			curr_pts = (avpkt.pts);
+			int out_linesize;
+			obuf_sz = av_rescale_rnd(swr_get_delay(swr, c->sample_rate) +
+						 frame->nb_samples, o_sr, c->sample_rate, AV_ROUND_UP);
+			if (obuf_sz > obuf_sz_max) {
+				lt_info("obuf_sz: %d old: %d\n", obuf_sz, obuf_sz_max);
+				av_free(obuf);
+				if (av_samples_alloc(&obuf, &out_linesize, o_ch,
+							frame->nb_samples, AV_SAMPLE_FMT_S16, 1) < 0) {
+					lt_info("av_samples_alloc failed\n");
+					av_free_packet(&avpkt);
+					break; /* while (thread_started) */
+				}
+				obuf_sz_max = obuf_sz;
+			}
+			obuf_sz = swr_convert(swr, &obuf, obuf_sz,
+					      (const uint8_t **)frame->extended_data, frame->nb_samples);
+			curr_pts = av_frame_get_best_effort_timestamp(frame);
 			apts = curr_pts;
 			lt_debug("aDec: pts 0x%" PRIx64 " %3f\n", curr_pts, curr_pts/90000.0);
-			int data_size = av_samples_get_buffer_size(NULL, c->channels,
+			int data_size = av_samples_get_buffer_size(&out_linesize, o_ch,
-					frame->nb_samples, c->sample_fmt, 1);
+								   obuf_sz, AV_SAMPLE_FMT_S16, 1);
 			packet = (packet_t *)malloc(sizeof(*packet));
 			packet->PTS = av_rescale_q(avpkt.pts, avfc->streams[0]->time_base, omx_timebase);
 			packet->DTS = -1;
 			packet->packetlength = data_size;
 			packet->packet = (uint8_t *)malloc(data_size);
-			memcpy(packet->packet, frame->data[0], data_size);
+			memcpy(packet->packet, obuf, data_size);
 			packet->buf = packet->packet;  /* This is what is free()ed */
 			codec_queue_add_item(&codecs.acodec, packet, MSG_PACKET);
 		}
@@ -419,6 +457,8 @@ void aDec::run()
 		av_free_packet(&avpkt);
 	}
 	lt_info("aDec: decoder loop exited\n");
+	swr_free(&swr);
+ out3:
 #if LIBAVCODEC_VERSION_INT >= (54 << 16 | 28 << 8)
 	avcodec_free_frame(&frame);
 #else