[alsa-plugins-freeworld] Updated to latest alsa git (libswresample)

commit 12c0d25ec4051d5db2be2c1dbfc5c4641a1268ff Author: Jaroslav Kysela <perex(a)perex.cz&gt; Date: Tue Nov 9 12:07:26 2021 +0100 Updated to latest alsa git (libswresample) Signed-off-by: Jaroslav Kysela <perex(a)perex.cz&gt; alsa-git.patch | 1312 +++++++++++++++++++++++++++++++++++++++++++ alsa-plugins-freeworld.spec | 7 +- 2 files changed, 1318 insertions(+), 1 deletion(-) --- diff --git a/alsa-git.patch b/alsa-git.patch new file mode 100644 index 0000000..876b275 --- /dev/null +++ b/alsa-git.patch @@ -0,0 +1,1312 @@ +diff --git a/a52/60-a52-encoder.conf b/a52/60-a52-encoder.conf +index 346c94f..99cc98e 100644 +--- a/a52/60-a52-encoder.conf ++++ b/a52/60-a52-encoder.conf +@@ -36,3 +36,10 @@ pcm.a52 { + description "Plugin to convert multichannel stream to A52 (AC3) bitstream" + } + } ++ ++# ++# A quick test command: ++# CARD=1 # replace with your IEC958 card ++# DEVICE=1 # replace with your IEC958 PCM device number ++# speaker-test -d -b 2000000 -c 6 -D"plug:{SLAVE=\"a52:${CARD},'hw:${CARD},${DEVICE}'\"}" ++# +diff --git a/a52/pcm_a52.c b/a52/pcm_a52.c +index b6a8f55..9da8897 100644 +--- a/a52/pcm_a52.c ++++ b/a52/pcm_a52.c +@@ -41,7 +41,12 @@ + #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53, 34, 0) + #include <libavutil/channel_layout.h> + #include <libavutil/mem.h> +-#define USE_AVCODEC_FRAME ++#define USE_AVCODEC_FRAME 1 ++#endif ++ ++#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(58, 91, 0) ++#include <libavcodec/packet.h> ++#define USE_AVCODEC_PACKET_ALLOC + #endif + + #if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(54, 0, 0) +@@ -62,6 +67,12 @@ + #define AV_CODEC_ID_AC3 CODEC_ID_AC3 + #endif + ++#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(56, 56, 0) ++#ifndef AV_INPUT_BUFFER_PADDING_SIZE ++#define AV_INPUT_BUFFER_PADDING_SIZE FF_INPUT_BUFFER_PADDING_SIZE ++#endif ++#endif ++ + #if LIBAVCODEC_VERSION_INT < 0x371c01 + #define av_frame_alloc avcodec_alloc_frame + #define av_frame_free avcodec_free_frame +@@ -72,20 +83,29 @@ struct a52_ctx { + snd_pcm_t *slave; + AVCodec *codec; + AVCodecContext *avctx; ++ snd_pcm_format_t src_format; ++ unsigned int src_sample_bits; ++ unsigned int src_sample_bytes; + snd_pcm_format_t format; + int av_format; + unsigned int channels; + unsigned int rate; + unsigned int bitrate; +- short *inbuf; ++ void *inbuf; + unsigned char *outbuf; ++ unsigned char *outbuf1; ++ unsigned char *outbuf2; + int outbuf_size; +- snd_pcm_uframes_t transfer; + int remain; + int filled; + unsigned int slave_period_size; + unsigned int slave_buffer_size; ++ snd_pcm_uframes_t pointer; ++ snd_pcm_uframes_t boundary; + snd_pcm_hw_params_t *hw_params; ++#ifdef USE_AVCODEC_PACKET_ALLOC ++ AVPacket *pkt; ++#endif + #ifdef USE_AVCODEC_FRAME + AVFrame *frame; + int is_planar; +@@ -98,70 +118,103 @@ struct a52_ctx { + #define use_planar(rec) 0 + #endif + +-#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53, 34, 0) ++#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(58, 91, 0) ++static int do_encode(struct a52_ctx *rec) ++{ ++ AVPacket *pkt = rec->pkt; ++ int ret; ++ ++ ret = avcodec_send_frame(rec->avctx, rec->frame); ++ if (ret < 0) ++ return -EINVAL; ++ ret = avcodec_receive_packet(rec->avctx, pkt); ++ if (ret < 0) ++ return -EINVAL; ++ ++ if (pkt->size > rec->outbuf_size - 8) ++ return -EINVAL; ++ memcpy(rec->outbuf1 + 8, pkt->data, pkt->size); ++ ++ return pkt->size; ++} ++#elif LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53, 34, 0) + static int do_encode(struct a52_ctx *rec) + { + AVPacket pkt = { +- .data = rec->outbuf + 8, ++ .data = rec->outbuf1 + 8, + .size = rec->outbuf_size - 8 + }; +- int got_frame; ++ int ret, got_frame; ++ ++ ret = avcodec_encode_audio2(rec->avctx, &pkt, rec->frame, &got_frame); ++ if (ret < 0) ++ return -EINVAL; + +- avcodec_encode_audio2(rec->avctx, &pkt, rec->frame, &got_frame); + return pkt.size; + } + #else + static int do_encode(struct a52_ctx *rec) + { +- return avcodec_encode_audio(rec->avctx, rec->outbuf + 8, +- rec->outbuf_size - 8, +- rec->inbuf); ++ int ret = avcodec_encode_audio(rec->avctx, rec->outbuf1 + 8, ++ rec->outbuf_size - 8, ++ rec->inbuf); ++ if (ret < 0) ++ return -EINVAL; ++ ++ return ret; + } + #endif + + /* convert the PCM data to A52 stream in IEC958 */ +-static void convert_data(struct a52_ctx *rec) ++static int convert_data(struct a52_ctx *rec) + { ++ unsigned char *buf; + int out_bytes = do_encode(rec); + +- rec->outbuf[0] = 0xf8; /* sync words */ +- rec->outbuf[1] = 0x72; +- rec->outbuf[2] = 0x4e; +- rec->outbuf[3] = 0x1f; +- rec->outbuf[4] = rec->outbuf[13] & 7; /* bsmod */ +- rec->outbuf[5] = 0x01; /* data type */ +- rec->outbuf[6] = ((out_bytes * 8) >> 8) & 0xff; +- rec->outbuf[7] = (out_bytes * 8) & 0xff; ++ if (out_bytes < 0) ++ return out_bytes; ++ ++ buf = rec->outbuf1; ++ buf[0] = 0xf8; /* sync words */ ++ buf[1] = 0x72; ++ buf[2] = 0x4e; ++ buf[3] = 0x1f; ++ buf[4] = buf[13] & 7; /* bsmod */ ++ buf[5] = 0x01; /* data type */ ++ buf[6] = ((out_bytes * 8) >> 8) & 0xff; ++ buf[7] = (out_bytes * 8) & 0xff; + /* swap bytes for little-endian 16bit */ +- if (rec->format == SND_PCM_FORMAT_S16_LE) +- swab(rec->outbuf, rec->outbuf, out_bytes + 8); +- memset(rec->outbuf + 8 + out_bytes, 0, ++ if (rec->format == SND_PCM_FORMAT_S16_LE) { ++ swab(rec->outbuf1, rec->outbuf2, out_bytes + 8); ++ buf = rec->outbuf2; ++ } ++ rec->outbuf = buf; ++ memset(buf + 8 + out_bytes, 0, + rec->outbuf_size - 8 - out_bytes); + rec->remain = rec->outbuf_size / 4; + rec->filled = 0; ++ ++ return 0; + } + + /* write pending encoded data to the slave pcm */ + static int write_out_pending(snd_pcm_ioplug_t *io, struct a52_ctx *rec) + { +- int err, ofs = (rec->avctx->frame_size - rec->remain) * 4; +- +- if (! rec->remain) +- return 0; ++ snd_pcm_sframes_t ret; ++ unsigned int ofs; + + while (rec->remain) { +- err = snd_pcm_writei(rec->slave, rec->outbuf + ofs, rec->remain); +- if (err < 0) { +- if (err == -EPIPE) ++ ofs = (rec->avctx->frame_size - rec->remain) * 4; ++ ret = snd_pcm_writei(rec->slave, rec->outbuf + ofs, rec->remain); ++ if (ret < 0) { ++ if (ret == -EPIPE) + io->state = SND_PCM_STATE_XRUN; +- if (err == -EAGAIN) ++ if (ret == -EAGAIN) + break; +- return err; +- } else if (! err) ++ return ret; ++ } else if (! ret) + break; +- if (err < rec->remain) +- ofs += (rec->remain - err) * 4; +- rec->remain -= err; ++ rec->remain -= ret; + } + return 0; + } +@@ -176,8 +229,8 @@ static void clear_remaining_planar_data(snd_pcm_ioplug_t *io) + unsigned int i; + + for (i = 0; i < io->channels; i++) +- memset(rec->frame->data[i] + rec->filled * 2, 0, +- (rec->avctx->frame_size - rec->filled) * 2); ++ memset(rec->frame->data[i] + rec->filled * rec->src_sample_bytes, 0, ++ (rec->avctx->frame_size - rec->filled) * rec->src_sample_bytes); + } + #else + #define clear_remaining_planar_data(io) /*NOP*/ +@@ -195,10 +248,12 @@ static int a52_drain(snd_pcm_ioplug_t *io) + if (use_planar(rec)) + clear_remaining_planar_data(io); + else { +- memset(rec->inbuf + rec->filled * io->channels, 0, +- (rec->avctx->frame_size - rec->filled) * io->channels * 2); ++ memset(rec->inbuf + rec->filled * io->channels * rec->src_sample_bytes, 0, ++ (rec->avctx->frame_size - rec->filled) * io->channels * rec->src_sample_bytes); + } +- convert_data(rec); ++ err = convert_data(rec); ++ if (err < 0) ++ return err; + } + err = write_out_pending(io, rec); + if (err < 0) +@@ -208,7 +263,8 @@ static int a52_drain(snd_pcm_ioplug_t *io) + } + + /* check whether the areas consist of a continuous interleaved stream */ +-static int check_interleaved(const snd_pcm_channel_area_t *areas, ++static int check_interleaved(struct a52_ctx *rec, ++ const snd_pcm_channel_area_t *areas, + unsigned int channels) + { + unsigned int ch; +@@ -218,8 +274,8 @@ static int check_interleaved(const snd_pcm_channel_area_t *areas, + + for (ch = 0; ch < channels; ch++) { + if (areas[ch].addr != areas[0].addr || +- areas[ch].first != ch * 16 || +- areas[ch].step != channels * 16) ++ areas[ch].first != ch * rec->src_sample_bits || ++ areas[ch].step != channels * rec->src_sample_bits) + return 0; + } + return 1; +@@ -237,8 +293,7 @@ static int fill_data(snd_pcm_ioplug_t *io, + { + struct a52_ctx *rec = io->private_data; + unsigned int len = rec->avctx->frame_size - rec->filled; +- short *src, *dst; +- unsigned int src_step; ++ void *_dst; + int err; + static unsigned int ch_index[3][6] = { + { 0, 1 }, +@@ -257,20 +312,23 @@ static int fill_data(snd_pcm_ioplug_t *io, + + /* If there are still frames left in outbuf, we can't + * accept a full a52 frame, because this would overwrite +- * the frames in outbuf. */ +- if (rec->remain && len) ++ * the frames in outbuf. This should not happen! The a52_pointer() ++ * callback should limit the transferred frames correctly. */ ++ if (rec->remain && len) { ++ SNDERR("fill data issue (remain is %i)", rec->remain); + len--; ++ } + + if (size > len) + size = len; + +- dst = rec->inbuf + rec->filled * io->channels; ++ _dst = rec->inbuf + rec->filled * io->channels * rec->src_sample_bytes; + if (!use_planar(rec) && interleaved) { +- memcpy(dst, areas->addr + offset * io->channels * 2, +- size * io->channels * 2); +- } else { +- unsigned int i, ch, dst_step; +- short *dst1; ++ memcpy(_dst, areas->addr + offset * io->channels * rec->src_sample_bytes, ++ size * io->channels * rec->src_sample_bytes); ++ } else if (rec->src_sample_bits == 16) { ++ unsigned int i, ch, src_step, dst_step; ++ short *src, *dst = _dst, *dst1; + + /* flatten copy to n-channel interleaved */ + dst_step = io->channels; +@@ -281,8 +339,8 @@ static int fill_data(snd_pcm_ioplug_t *io, + (ap->first + offset * ap->step) / 8); + + #ifdef USE_AVCODEC_FRAME +- if (use_planar(rec)) { +- memcpy(rec->frame->data[ch], src, size * 2); ++ if (use_planar(rec) && !interleaved) { ++ memcpy(rec->frame->data[ch] + rec->filled * 2, src, size * 2); + continue; + } + #endif +@@ -294,10 +352,40 @@ static int fill_data(snd_pcm_ioplug_t *io, + dst1 += dst_step; + } + } ++ } else if (rec->src_sample_bits == 32) { ++ unsigned int i, ch, src_step, dst_step; ++ int *src, *dst = _dst, *dst1; ++ ++ /* flatten copy to n-channel interleaved */ ++ dst_step = io->channels; ++ for (ch = 0; ch < io->channels; ch++, dst++) { ++ const snd_pcm_channel_area_t *ap; ++ ap = &areas[ch_index[io->channels / 2 - 1][ch]]; ++ src = (int *)(ap->addr + ++ (ap->first + offset * ap->step) / 8); ++ ++#ifdef USE_AVCODEC_FRAME ++ if (use_planar(rec) && !interleaved) { ++ memcpy(rec->frame->data[ch] + rec->filled * 4, src, size * 4); ++ continue; ++ } ++#endif ++ dst1 = dst; ++ src_step = ap->step / 32; /* in word */ ++ for (i = 0; i < size; i++) { ++ *dst1 = *src; ++ src += src_step; ++ dst1 += dst_step; ++ } ++ } ++ } else { ++ return -EIO; + } + rec->filled += size; + if (rec->filled == rec->avctx->frame_size) { +- convert_data(rec); ++ err = convert_data(rec); ++ if (err < 0) ++ return err; + write_out_pending(io, rec); + } + return (int)size; +@@ -314,7 +402,7 @@ static snd_pcm_sframes_t a52_transfer(snd_pcm_ioplug_t *io, + struct a52_ctx *rec = io->private_data; + snd_pcm_sframes_t result = 0; + int err = 0; +- int interleaved = check_interleaved(areas, io->channels); ++ int interleaved = check_interleaved(rec, areas, io->channels); + + do { + err = fill_data(io, areas, offset, size, interleaved); +@@ -323,7 +411,8 @@ static snd_pcm_sframes_t a52_transfer(snd_pcm_ioplug_t *io, + offset += (unsigned int)err; + size -= (unsigned int)err; + result += err; +- rec->transfer += err; ++ rec->pointer += err; ++ rec->pointer %= rec->boundary; + } while (size); + return result > 0 ? result : err; + } +@@ -336,7 +425,7 @@ static snd_pcm_sframes_t a52_transfer(snd_pcm_ioplug_t *io, + static snd_pcm_sframes_t a52_pointer(snd_pcm_ioplug_t *io) + { + struct a52_ctx *rec = io->private_data; +- snd_pcm_sframes_t avail; ++ snd_pcm_sframes_t avail, delay; + snd_pcm_state_t state; + + state = snd_pcm_state(rec->slave); +@@ -345,31 +434,31 @@ static snd_pcm_sframes_t a52_pointer(snd_pcm_ioplug_t *io) + case SND_PCM_STATE_DRAINING: + break; + case SND_PCM_STATE_XRUN: +- case SND_PCM_STATE_SUSPENDED: + return -EPIPE; ++ case SND_PCM_STATE_SUSPENDED: ++ return -ESTRPIPE; + default: + return 0; + } + +- avail = 0; +- + /* Write what we have from outbuf. */ + write_out_pending(io, rec); + +- /* If there is anything remaining in outbuf, we can't +- * accept any full packets. */ +- if (rec->remain == 0) +- { +- /* Round the slave frames to multiples of the packet size. */ +- avail += (snd_pcm_avail_update(rec->slave) / rec->avctx->frame_size) * rec->avctx->frame_size; +- } +- ++ avail = snd_pcm_avail(rec->slave); + if (avail < 0) +- avail = 0; +- else if (avail >= io->buffer_size) +- avail = io->buffer_size - 1; ++ return avail; + +- return (io->appl_ptr + avail) % io->buffer_size; ++ /* get buffer delay without additional FIFO information */ ++ delay = rec->slave_buffer_size - avail; ++ while (delay < 0) ++ delay += rec->slave_buffer_size; ++ ++ avail = rec->pointer - delay - rec->remain - rec->filled; ++#ifdef SND_PCM_IOPLUG_FLAG_BOUNDARY_WA ++ return avail % rec->boundary; ++#else ++ return avail % io->buffer_size; ++#endif + } + + /* set up the fixed parameters of slave PCM hw_parmas */ +@@ -462,6 +551,25 @@ static int a52_hw_free(snd_pcm_ioplug_t *io) + return snd_pcm_hw_free(rec->slave); + } + ++/* ++ * dump callback ++ */ ++static void a52_dump(snd_pcm_ioplug_t *io, snd_output_t *out) ++{ ++ struct a52_ctx *rec = io->private_data; ++ snd_pcm_t *pcm = io->pcm; ++ ++ snd_output_printf(out, "%s\n", io->name); ++ snd_output_printf(out, "Its setup is:\n"); ++ snd_pcm_dump_setup(pcm, out); ++ snd_output_printf(out, " %-13s: %s\n", "av_format", av_get_sample_fmt_name(rec->av_format)); ++ snd_output_printf(out, " %-13s: %i\n", "av_frame_size", rec->avctx ? rec->avctx->frame_size : -1); ++ snd_output_printf(out, " %-13s: %i\n", "remain", rec->remain); ++ snd_output_printf(out, " %-13s: %i\n", "filled", rec->filled); ++ snd_output_printf(out, "Slave: "); ++ snd_pcm_dump(rec->slave, out); ++} ++ + /* + * sw_params callback + * +@@ -476,6 +584,7 @@ static int a52_sw_params(snd_pcm_ioplug_t *io, snd_pcm_sw_params_t *params) + + snd_pcm_sw_params_get_avail_min(params, &avail_min); + snd_pcm_sw_params_get_start_threshold(params, &start_threshold); ++ snd_pcm_sw_params_get_boundary(params, &rec->boundary); + + len = avail_min; + len += (int)rec->slave_buffer_size - (int)io->buffer_size; +@@ -525,21 +634,27 @@ static void a52_free(struct a52_ctx *rec) + } + + #ifdef USE_AVCODEC_FRAME +- if (rec->frame) { ++#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(58, 91, 0) ++ if (rec->frame) + av_freep(&rec->frame->data[0]); +- rec->inbuf = NULL; +- } ++#endif + #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(54, 28, 0) + av_frame_free(&rec->frame); + #else + av_freep(&rec->frame); + #endif +-#endif +- ++#else /* USE_AVCODEC_FRAME */ + free(rec->inbuf); + rec->inbuf = NULL; +- free(rec->outbuf); +- rec->outbuf = NULL; ++#endif /* USE_AVCODEC_FRAME */ ++ ++#ifdef USE_AVCODEC_PACKET_ALLOC ++ av_packet_free(&rec->pkt); ++#endif ++ free(rec->outbuf2); ++ rec->outbuf2 = NULL; ++ free(rec->outbuf1); ++ rec->outbuf1 = NULL; + } + + /* +@@ -577,14 +692,22 @@ static int alloc_input_buffer(snd_pcm_ioplug_t *io) + rec->frame = av_frame_alloc(); + if (!rec->frame) + return -ENOMEM; ++ rec->frame->nb_samples = rec->avctx->frame_size; ++#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(58, 91, 0) ++ rec->frame->format = rec->avctx->sample_fmt; ++ rec->frame->channels = rec->avctx->channels; ++ rec->frame->channel_layout = rec->avctx->channel_layout; ++ if (av_frame_get_buffer(rec->frame, 0)) ++ return -ENOMEM; ++#else + if (av_samples_alloc(rec->frame->data, rec->frame->linesize, + io->channels, rec->avctx->frame_size, + rec->avctx->sample_fmt, 0) < 0) + return -ENOMEM; +- rec->frame->nb_samples = rec->avctx->frame_size; +- rec->inbuf = (short *)rec->frame->data[0]; ++#endif ++ rec->inbuf = rec->frame->data[0]; + #else +- rec->inbuf = malloc(rec->avctx->frame_size * 2 * io->channels); ++ rec->inbuf = malloc(rec->avctx->frame_size * io->channels * rec->src_sample_bytes); + #endif + if (!rec->inbuf) + return -ENOMEM; +@@ -622,15 +745,28 @@ static int a52_prepare(snd_pcm_ioplug_t *io) + if (err < 0) + return -EINVAL; + ++#ifdef USE_AVCODEC_PACKET_ALLOC ++ rec->pkt = av_packet_alloc(); ++ if (!rec->pkt) ++ return -ENOMEM; ++#endif ++ + rec->outbuf_size = rec->avctx->frame_size * 4; +- rec->outbuf = malloc(rec->outbuf_size); +- if (! rec->outbuf) ++ rec->outbuf1 = malloc(rec->outbuf_size + AV_INPUT_BUFFER_PADDING_SIZE); ++ if (! rec->outbuf1) + return -ENOMEM; ++ memset(rec->outbuf1 + rec->outbuf_size, 0, AV_INPUT_BUFFER_PADDING_SIZE); ++ ++ if (rec->format == SND_PCM_FORMAT_S16_LE) { ++ rec->outbuf2 = malloc(rec->outbuf_size); ++ if ( !rec->outbuf2) ++ return -ENOMEM; ++ } + + if (alloc_input_buffer(io)) + return -ENOMEM; + +- rec->transfer = 0; ++ rec->pointer = 0; + rec->remain = 0; + rec->filled = 0; + +@@ -669,10 +805,9 @@ static int a52_close(snd_pcm_ioplug_t *io) + snd_pcm_t *slave = rec->slave; + + a52_free(rec); +- if (slave) { +- rec->slave = NULL; ++ free(rec); ++ if (slave) + return snd_pcm_close(slave); +- } + return 0; + } + +@@ -737,6 +872,7 @@ static snd_pcm_ioplug_callback_t a52_ops = { + .close = a52_close, + .hw_params = a52_hw_params, + .hw_free = a52_hw_free, ++ .dump = a52_dump, + .sw_params = a52_sw_params, + .prepare = a52_prepare, + .drain = a52_drain, +@@ -773,8 +909,20 @@ static int a52_set_hw_constraint(struct a52_ctx *rec) + SND_PCM_ACCESS_MMAP_NONINTERLEAVED, + SND_PCM_ACCESS_RW_NONINTERLEAVED + }; +- unsigned int formats[] = { SND_PCM_FORMAT_S16 }; +- int err; ++ static struct format { ++ int av; ++ snd_pcm_format_t alib; ++ } formats[] = { ++ { .av = AV_SAMPLE_FMT_S16, .alib = SND_PCM_FORMAT_S16 }, ++#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52, 95, 0) && USE_AVCODEC_FRAME ++ { .av = AV_SAMPLE_FMT_S16P, .alib = SND_PCM_FORMAT_S16 }, ++ { .av = AV_SAMPLE_FMT_S32, .alib = SND_PCM_FORMAT_S32 }, ++ { .av = AV_SAMPLE_FMT_S32P, .alib = SND_PCM_FORMAT_S32 }, ++ { .av = AV_SAMPLE_FMT_FLTP, .alib = SND_PCM_FORMAT_FLOAT } ++#endif ++ }; ++ int err, dir; ++ unsigned int i, fmt; + snd_pcm_uframes_t buffer_max; + unsigned int period_bytes, max_periods; + +@@ -791,8 +939,22 @@ static int a52_set_hw_constraint(struct a52_ctx *rec) + if (err < 0) + return err; + ++ rec->src_format = SND_PCM_FORMAT_UNKNOWN; ++ for (i = 0; i < ARRAY_SIZE(formats); i++) ++ if (formats[i].av == rec->av_format) { ++ rec->src_format = formats[i].alib; ++ break; ++ } ++ if (rec->src_format == SND_PCM_FORMAT_UNKNOWN) { ++ SNDERR("A/V format '%s' is not supported", av_get_sample_fmt_name(rec->av_format)); ++ return -EINVAL; ++ } ++ fmt = rec->src_format; ++ rec->src_sample_bits = snd_pcm_format_physical_width(rec->src_format); ++ rec->src_sample_bytes = rec->src_sample_bits / 8; ++ + if ((err = snd_pcm_ioplug_set_param_list(&rec->io, SND_PCM_IOPLUG_HW_FORMAT, +- ARRAY_SIZE(formats), formats)) < 0 || ++ 1, &fmt)) < 0 || + (err = snd_pcm_ioplug_set_param_minmax(&rec->io, SND_PCM_IOPLUG_HW_CHANNELS, + rec->channels, rec->channels)) < 0 || + (err = snd_pcm_ioplug_set_param_minmax(&rec->io, SND_PCM_IOPLUG_HW_RATE, +@@ -803,8 +965,11 @@ static int a52_set_hw_constraint(struct a52_ctx *rec) + return err; + + snd_pcm_hw_params_get_buffer_size_max(rec->hw_params, &buffer_max); ++ dir = -1; ++ snd_pcm_hw_params_get_periods_max(rec->hw_params, &max_periods, &dir); + period_bytes = A52_FRAME_SIZE * 2 * rec->channels; +- max_periods = buffer_max / A52_FRAME_SIZE; ++ if (buffer_max / A52_FRAME_SIZE < max_periods) ++ max_periods = buffer_max / A52_FRAME_SIZE; + + if ((err = snd_pcm_ioplug_set_param_minmax(&rec->io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, + period_bytes, period_bytes)) < 0 || +@@ -824,11 +989,12 @@ SND_PCM_PLUGIN_DEFINE_FUNC(a52) + int err; + const char *card = NULL; + const char *pcm_string = NULL; ++ const char *avcodec = NULL; + unsigned int rate = 48000; + unsigned int bitrate = 448; + unsigned int channels = 6; + snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE; +- char devstr[128], tmpcard[8]; ++ char devstr[128], tmpcard[16]; + struct a52_ctx *rec; + + if (stream != SND_PCM_STREAM_PLAYBACK) { +@@ -921,6 +1087,16 @@ SND_PCM_PLUGIN_DEFINE_FUNC(a52) + } + continue; + } ++ if (strcmp(id, "avcodec") == 0) { ++ const char *str; ++ err = snd_config_get_string(n, &str); ++ if (err < 0) { ++ SNDERR("invalid type for %s", id); ++ return -EINVAL; ++ } ++ avcodec = str; ++ continue; ++ } + SNDERR("Unknown field %s", id); + return -EINVAL; + } +@@ -931,6 +1107,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(a52) + return -ENOMEM; + } + ++ rec->src_format = SND_PCM_FORMAT_UNKNOWN; + rec->rate = rate; + rec->bitrate = bitrate; + rec->channels = channels; +@@ -939,11 +1116,17 @@ SND_PCM_PLUGIN_DEFINE_FUNC(a52) + #ifndef USE_AVCODEC_FRAME + avcodec_init(); + #endif ++#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(58, 91, 0) + avcodec_register_all(); ++#endif + +- rec->codec = avcodec_find_encoder_by_name("ac3_fixed"); +- if (rec->codec == NULL) +- rec->codec = avcodec_find_encoder_by_name("ac3"); ++ if (avcodec) { ++ rec->codec = avcodec_find_encoder_by_name(avcodec); ++ } else { ++ rec->codec = avcodec_find_encoder_by_name("ac3_fixed"); ++ if (rec->codec == NULL) ++ rec->codec = avcodec_find_encoder_by_name("ac3"); ++ } + if (rec->codec == NULL) + rec->codec = avcodec_find_encoder(AV_CODEC_ID_AC3); + if (rec->codec == NULL) { +@@ -954,7 +1137,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(a52) + + if (! pcm_string || pcm_string[0] == '\0') { + snprintf(devstr, sizeof(devstr), +- "iec958:{AES0 0x%x AES1 0x%x AES2 0x%x AES3 0x%x %s%s}", ++ "iec958:{AES0 0x%x AES1 0x%x AES2 0x%x AES3 0x%x%s%s}", + IEC958_AES0_CON_EMPHASIS_NONE | IEC958_AES0_NONAUDIO | + IEC958_AES0_CON_NOT_COPYRIGHT, + IEC958_AES1_CON_ORIGINAL | IEC958_AES1_CON_PCM_CODER, +@@ -980,6 +1163,9 @@ SND_PCM_PLUGIN_DEFINE_FUNC(a52) + rec->io.mmap_rw = 0; + rec->io.callback = &a52_ops; + rec->io.private_data = rec; ++#ifdef SND_PCM_IOPLUG_FLAG_BOUNDARY_WA ++ rec->io.flags = SND_PCM_IOPLUG_FLAG_BOUNDARY_WA; ++#endif + #ifdef USE_AVCODEC_FRAME + rec->av_format = rec->codec->sample_fmts[0]; + rec->is_planar = av_sample_fmt_is_planar(rec->av_format); +@@ -993,7 +1179,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(a52) + + if ((err = a52_set_hw_constraint(rec)) < 0) { + snd_pcm_ioplug_delete(&rec->io); +- goto error; ++ return err; + } + + *pcmp = rec->io.pcm; +diff --git a/configure.ac b/configure.ac +index d5fe529..860daa9 100644 +--- a/configure.ac ++++ b/configure.ac +@@ -93,7 +93,7 @@ AC_ARG_ENABLE([libav], + AS_HELP_STRING([--disable-libav], [Do not build plugins depending on libav/ffmpeg (a52,lavrate...)])) + + if test "x$enable_libav" != "xno"; then +- PKG_CHECK_MODULES(LIBAV, [libavcodec libavutil libavresample], [HAVE_LIBAV=yes], [HAVE_LIBAV=no]) ++ PKG_CHECK_MODULES(LIBAV, [libavcodec libavutil libswresample], [HAVE_LIBAV=yes], [HAVE_LIBAV=no]) + fi + + if test "x$HAVE_LIBAV" = "xno"; then +diff --git a/doc/lavrate.txt b/doc/lavrate.txt +index 6575183..fa6bbb0 100644 +--- a/doc/lavrate.txt ++++ b/doc/lavrate.txt +@@ -2,7 +2,7 @@ Rate Converter Plugin Using libavresample + =========================================0 + + The plugin in rate-lavr subdirectory is an external rate converter using +-libavresample library. You can use this rate converter plugin by defining a ++libswresample library. You can use this rate converter plugin by defining a + rate PCM with "converter" parameter, such as: + + pcm.my_rate { +@@ -16,7 +16,7 @@ The plug plugin has also a similar field, "rate_converter". + Or, more easily, define a global variable "defaults.pcm.rate_converter", + which is used as the default converter type by plug and rate plugins: + +- defaults.pcm.rate_converter "lavcrate" ++ defaults.pcm.rate_converter "lavrate" + + Write the above in your ~/.asoundrc or /etc/asound.conf. + +diff --git a/jack/Makefile.am b/jack/Makefile.am +index 7801194..117a5c9 100644 +--- a/jack/Makefile.am ++++ b/jack/Makefile.am +@@ -12,7 +12,7 @@ AM_CFLAGS = -Wall -g @ALSA_CFLAGS@ @JACK_CFLAGS@ + AM_LDFLAGS = -module -avoid-version -export-dynamic -no-undefined $(LDFLAGS_NOUNDEFINED) + + libasound_module_pcm_jack_la_SOURCES = pcm_jack.c +-libasound_module_pcm_jack_la_LIBADD = @ALSA_LIBS@ @JACK_LIBS@ ++libasound_module_pcm_jack_la_LIBADD = @ALSA_LIBS@ @JACK_LIBS@ -lpthread + + include ../install-hooks.am + +diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c +index 26674fa..436aefe 100644 +--- a/jack/pcm_jack.c ++++ b/jack/pcm_jack.c +@@ -28,6 +28,7 @@ + #include <jack/jack.h> + #include <alsa/asoundlib.h> + #include <alsa/pcm_external.h> ++#include <pthread.h> + + #define MAX_PERIODS_MULTIPLE 64 + +@@ -44,6 +45,8 @@ typedef struct { + + int fd; + int activated; /* jack is activated? */ ++ pthread_mutex_t running_mutex; ++ int running; /* jack is running? */ + + snd_pcm_jack_port_list_t **port_names; + unsigned int num_ports; +@@ -159,6 +162,7 @@ static void snd_pcm_jack_free(snd_pcm_jack_t *jack) + free(jack->port_names); + jack->port_names = NULL; + } ++ pthread_mutex_destroy (&jack->running_mutex); + if (jack->fd >= 0) + close(jack->fd); + if (jack->io.poll_fd >= 0) +@@ -207,6 +211,18 @@ snd_pcm_jack_process_cb(jack_nframes_t nframes, snd_pcm_ioplug_t *io) + snd_pcm_jack_t *jack = io->private_data; + snd_pcm_uframes_t xfer = 0; + unsigned int channel; ++ ++ if (pthread_mutex_trylock (&jack->running_mutex) == EBUSY) { ++ /* Note that locking should only ever fail if ++ * snd_pcm_jack_start or snd_pcm_jack_stop is called at the ++ * same time, in which case dropping the current buffer is not ++ * an issue. */ ++ return 0; ++ } ++ if (!jack->running) { ++ pthread_mutex_unlock (&jack->running_mutex); ++ return 0; ++ } + + for (channel = 0; channel < io->channels; channel++) { + jack->areas[channel].addr = +@@ -268,6 +284,8 @@ snd_pcm_jack_process_cb(jack_nframes_t nframes, snd_pcm_ioplug_t *io) + + pcm_poll_unblock_check(io); /* unblock socket for polling if needed */ + ++ pthread_mutex_unlock (&jack->running_mutex); ++ + return 0; + } + +@@ -365,7 +383,10 @@ static int snd_pcm_jack_prepare(snd_pcm_ioplug_t *io) + + static int snd_pcm_jack_start(snd_pcm_ioplug_t *io) + { +- (void)io; ++ snd_pcm_jack_t *jack = io->private_data; ++ pthread_mutex_lock (&jack->running_mutex); ++ jack->running = 1; ++ pthread_mutex_unlock (&jack->running_mutex); + /* + * Since the processing of jack_activate() and jack_connect() take a + * while longer, snd_pcm_jack_start() was blocked. +@@ -382,7 +403,10 @@ static int snd_pcm_jack_start(snd_pcm_ioplug_t *io) + + static int snd_pcm_jack_stop(snd_pcm_ioplug_t *io) + { +- (void)io; ++ snd_pcm_jack_t *jack = io->private_data; ++ pthread_mutex_lock (&jack->running_mutex); ++ jack->running = 0; ++ pthread_mutex_unlock (&jack->running_mutex); + return 0; + } + +@@ -548,6 +572,8 @@ static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, + if (!jack) + return -ENOMEM; + ++ pthread_mutex_init (&jack->running_mutex, NULL); ++ + jack->fd = -1; + jack->io.poll_fd = -1; + jack->use_period_alignment = use_period_alignment; +diff --git a/rate-lav/rate_lavrate.c b/rate-lav/rate_lavrate.c +index 2b992c5..f78eea5 100644 +--- a/rate-lav/rate_lavrate.c ++++ b/rate-lav/rate_lavrate.c +@@ -1,5 +1,5 @@ + /* +- * Rate converter plugin using libavresample ++ * Rate converter plugin using libswresample + * Copyright (c) 2014 by Anton Khirnov + * + * This library is free software; you can redistribute it and/or +@@ -17,7 +17,7 @@ + #include <alsa/asoundlib.h> + #include <alsa/pcm_rate.h> + +-#include <libavresample/avresample.h> ++#include <libswresample/swresample.h> + #include <libavutil/channel_layout.h> + #include <libavutil/opt.h> + #include <libavutil/mathematics.h> +@@ -25,15 +25,15 @@ + + + static unsigned int filter_size = 16; +-static unsigned int phase_shift = 10; /* auto-adjusts */ +-static double cutoff = 0; /* auto-adjusts */ + + struct rate_src { +- AVAudioResampleContext *avr; ++ SwrContext *avr; + + unsigned int in_rate; + unsigned int out_rate; + unsigned int channels; ++ ++ unsigned int version; + }; + + static snd_pcm_uframes_t input_frames(void *obj ATTRIBUTE_UNUSED, +@@ -51,51 +51,67 @@ static snd_pcm_uframes_t output_frames(void *obj ATTRIBUTE_UNUSED, + static void pcm_src_free(void *obj) + { + struct rate_src *rate = obj; +- avresample_free(&rate->avr); ++ swr_free(&rate->avr); ++} ++ ++static int to_av_format(snd_pcm_format_t f) ++{ ++ switch (f) { ++ case SND_PCM_FORMAT_FLOAT: ++ return AV_SAMPLE_FMT_FLT; ++ case SND_PCM_FORMAT_U8: ++ return AV_SAMPLE_FMT_U8; ++ case SND_PCM_FORMAT_S16: ++ return AV_SAMPLE_FMT_S16; ++ case SND_PCM_FORMAT_S32: ++ default: ++ return AV_SAMPLE_FMT_S32; ++ } ++} ++ ++static int support_multi_format(struct rate_src *rate) ++{ ++#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010003 ++ return rate->version >= 0x010003; ++#else ++ return 0; ++#endif + } + + static int pcm_src_init(void *obj, snd_pcm_rate_info_t *info) + { + struct rate_src *rate = obj; +- int i, ir, or; ++ int fmt; + + if (!rate->avr || rate->channels != info->channels) { + int ret; + + pcm_src_free(rate); + rate->channels = info->channels; +- ir = rate->in_rate = info->in.rate; +- or = rate->out_rate = info->out.rate; +- i = av_gcd(or, ir); +- if (or > ir) { +- phase_shift = or/i; +- } else { +- phase_shift = ir/i; +- } +- if (cutoff <= 0.0) { +- cutoff = 1.0 - 1.0/filter_size; +- if (cutoff < 0.80) +- cutoff = 0.80; +- } ++ rate->in_rate = info->in.rate; ++ rate->out_rate = info->out.rate; + +- rate->avr = avresample_alloc_context(); ++ rate->avr = swr_alloc(); + if (!rate->avr) + return -ENOMEM; + +- av_opt_set_int(rate->avr, "in_sample_rate", info->in.rate, 0); +- av_opt_set_int(rate->avr, "out_sample_rate", info->out.rate, 0); +- av_opt_set_int(rate->avr, "in_sample_format", AV_SAMPLE_FMT_S16, 0); +- av_opt_set_int(rate->avr, "out_sample_format", AV_SAMPLE_FMT_S16, 0); +- av_opt_set_int(rate->avr, "in_channel_layout", av_get_default_channel_layout(rate->channels), 0); +- av_opt_set_int(rate->avr, "out_channel_layout", av_get_default_channel_layout(rate->channels), 0); +- +- av_opt_set_int(rate->avr, "filter_size", filter_size, 0); +- av_opt_set_int(rate->avr, "phase_shift", phase_shift, 0); +- av_opt_set_double(rate->avr, "cutoff", cutoff, 0); +- +- ret = avresample_open(rate->avr); ++ av_opt_set_channel_layout(rate->avr, "in_channel_layout", ++ av_get_default_channel_layout(rate->channels), 0); ++ av_opt_set_channel_layout(rate->avr, "out_channel_layout", ++ av_get_default_channel_layout(rate->channels), 0); ++ av_opt_set_int(rate->avr, "in_sample_rate", rate->in_rate, 0); ++ av_opt_set_int(rate->avr, "out_sample_rate", rate->out_rate, 0); ++ fmt = support_multi_format(rate) ? info->in.format : SND_PCM_FORMAT_S16; ++ av_opt_set_sample_fmt(rate->avr, "in_sample_fmt", ++ to_av_format(fmt), 0); ++ fmt = support_multi_format(rate) ? info->out.format : SND_PCM_FORMAT_S16; ++ av_opt_set_sample_fmt(rate->avr, "out_sample_fmt", ++ to_av_format(fmt), 0); ++ ++ ret = swr_init(rate->avr); + if (ret < 0) { +- avresample_free(&rate->avr); ++ SNDERR("sw_init() error %d\n", ret); ++ swr_free(&rate->avr); + return -EINVAL; + } + } +@@ -118,26 +134,51 @@ static void pcm_src_reset(void *obj) + + if (rate->avr) { + #if 0 +- avresample_close(rate->avr); +- avresample_open(rate->avr); ++ swr_free(rate->avr); ++ swr_init(rate->avr); + #endif + } + } + ++static void do_convert(struct rate_src *rate, ++ void *dst, unsigned int dst_frames, ++ const void *src, unsigned int src_frames) ++{ ++ unsigned int total_in = swr_get_delay(rate->avr, rate->in_rate) + src_frames; ++ ++ swr_convert(rate->avr, (uint8_t **)&dst, dst_frames, ++ (const uint8_t **)&src, src_frames); ++ ++ swr_set_compensation(rate->avr, ++ total_in - src_frames > filter_size ? 0 : 1, ++ src_frames); ++} ++ ++#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010003 ++static void pcm_src_convert(void *obj, ++ const snd_pcm_channel_area_t *dst_areas, ++ snd_pcm_uframes_t dst_offset, ++ unsigned int dst_frames, ++ const snd_pcm_channel_area_t *src_areas, ++ snd_pcm_uframes_t src_offset, ++ unsigned int src_frames) ++{ ++ struct rate_src *rate = obj; ++ const void *src = snd_pcm_channel_area_addr(src_areas, src_offset); ++ void *dst = snd_pcm_channel_area_addr(dst_areas, dst_offset); ++ ++ do_convert(rate, dst, dst_frames, src, src_frames); ++} ++#endif ++ + static void pcm_src_convert_s16(void *obj, int16_t *dst, + unsigned int dst_frames, + const int16_t *src, + unsigned int src_frames) + { + struct rate_src *rate = obj; +- int chans = rate->channels; +- unsigned int total_in = avresample_get_delay(rate->avr) + src_frames; + +- avresample_convert(rate->avr, (uint8_t **)&dst, dst_frames * chans * 2, dst_frames, +- (uint8_t **)&src, src_frames * chans * 2, src_frames); +- +- avresample_set_compensation(rate->avr, +- total_in - src_frames > filter_size ? 0 : 1, src_frames); ++ do_convert(rate, dst, dst_frames, src, src_frames); + } + + static void pcm_src_close(void *obj) +@@ -156,7 +197,21 @@ static int get_supported_rates(void *obj ATTRIBUTE_UNUSED, + + static void dump(void *obj ATTRIBUTE_UNUSED, snd_output_t *out) + { +- snd_output_printf(out, "Converter: libavr\n"); ++ snd_output_printf(out, "Converter: libswresample\n"); ++} ++#endif ++ ++#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010003 ++static int get_supported_formats(void *obj, uint64_t *in_formats, ++ uint64_t *out_formats, ++ unsigned int *flags) ++{ ++ *in_formats = *out_formats = ++ (1ULL << SND_PCM_FORMAT_U8) | ++ (1ULL << SND_PCM_FORMAT_S16) | ++ (1ULL << SND_PCM_FORMAT_S32); ++ *flags = SND_PCM_RATE_FLAG_INTERLEAVED; ++ return 0; + } + #endif + +@@ -166,6 +221,9 @@ static snd_pcm_rate_ops_t pcm_src_ops = { + .free = pcm_src_free, + .reset = pcm_src_reset, + .adjust_pitch = pcm_src_adjust_pitch, ++#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010003 ++ .convert = pcm_src_convert, ++#endif + .convert_s16 = pcm_src_convert_s16, + .input_frames = input_frames, + .output_frames = output_frames, +@@ -174,31 +232,35 @@ static snd_pcm_rate_ops_t pcm_src_ops = { + .get_supported_rates = get_supported_rates, + .dump = dump, + #endif ++#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010003 ++ .get_supported_formats = get_supported_formats, ++#endif + }; + + int pcm_src_open(unsigned int version, void **objp, snd_pcm_rate_ops_t *ops) +- + { + struct rate_src *rate; + +-#if SND_PCM_RATE_PLUGIN_VERSION < 0x010002 +- if (version != SND_PCM_RATE_PLUGIN_VERSION) { +- fprintf(stderr, "Invalid rate plugin version %x\n", version); +- return -EINVAL; +- } +-#endif + rate = calloc(1, sizeof(*rate)); + if (!rate) + return -ENOMEM; + + *objp = rate; + rate->avr = NULL; ++ rate->version = version; + #if SND_PCM_RATE_PLUGIN_VERSION >= 0x010002 +- if (version == 0x010001) ++ if (version == 0x010001) { + memcpy(ops, &pcm_src_ops, sizeof(snd_pcm_rate_old_ops_t)); +- else ++ return 0; ++ } ++#endif ++#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010003 ++ if (version == 0x010002) { ++ memcpy(ops, &pcm_src_ops, sizeof(snd_pcm_rate_v2_ops_t)); ++ return 0; ++ } + #endif +- *ops = pcm_src_ops; ++ *ops = pcm_src_ops; + return 0; + } + +diff --git a/rate/rate_samplerate.c b/rate/rate_samplerate.c +index 100d6f2..dd0a416 100644 +--- a/rate/rate_samplerate.c ++++ b/rate/rate_samplerate.c +@@ -40,9 +40,12 @@ + #include <alsa/pcm_rate.h> + + struct rate_src { ++ unsigned int version; + double ratio; + int converter; + unsigned int channels; ++ int in_int; ++ int out_int; + float *src_buf; + float *dst_buf; + SRC_STATE *state; +@@ -109,6 +112,13 @@ static int pcm_src_init(void *obj, snd_pcm_rate_info_t *info) + rate->data.src_ratio = rate->ratio; + rate->data.end_of_input = 0; + ++#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010003 ++ if (rate->version >= 0x010003) { ++ rate->in_int = info->in.format == SND_PCM_FORMAT_S32; ++ rate->out_int = info->out.format == SND_PCM_FORMAT_S32; ++ } ++#endif ++ + return 0; + } + +@@ -128,24 +138,56 @@ static void pcm_src_reset(void *obj) + src_reset(rate->state); + } + +-static void pcm_src_convert_s16(void *obj, int16_t *dst, unsigned int dst_frames, +- const int16_t *src, unsigned int src_frames) ++static void do_convert(struct rate_src *rate, ++ void *dst, unsigned int dst_frames, ++ const void *src, unsigned int src_frames) + { +- struct rate_src *rate = obj; + unsigned int ofs; + + rate->data.input_frames = src_frames; + rate->data.output_frames = dst_frames; + rate->data.end_of_input = 0; + +- src_short_to_float_array(src, rate->src_buf, src_frames * rate->channels); ++ if (rate->in_int) ++ src_int_to_float_array(src, rate->src_buf, src_frames * rate->channels); ++ else ++ src_short_to_float_array(src, rate->src_buf, src_frames * rate->channels); + src_process(rate->state, &rate->data); + if (rate->data.output_frames_gen < dst_frames) + ofs = dst_frames - rate->data.output_frames_gen; + else + ofs = 0; +- src_float_to_short_array(rate->dst_buf, dst + ofs * rate->channels, +- rate->data.output_frames_gen * rate->channels); ++ if (rate->out_int) ++ src_float_to_int_array(rate->dst_buf, dst + ofs * rate->channels * 4, ++ rate->data.output_frames_gen * rate->channels); ++ else ++ src_float_to_short_array(rate->dst_buf, dst + ofs * rate->channels * 2, ++ rate->data.output_frames_gen * rate->channels); ++} ++ ++#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010003 ++static void pcm_src_convert(void *obj, ++ const snd_pcm_channel_area_t *dst_areas, ++ snd_pcm_uframes_t dst_offset, ++ unsigned int dst_frames, ++ const snd_pcm_channel_area_t *src_areas, ++ snd_pcm_uframes_t src_offset, ++ unsigned int src_frames) ++{ ++ struct rate_src *rate = obj; ++ const void *src = snd_pcm_channel_area_addr(src_areas, src_offset); ++ void *dst = snd_pcm_channel_area_addr(dst_areas, dst_offset); ++ ++ do_convert(rate, dst, dst_frames, src, src_frames); ++} ++#endif ++ ++static void pcm_src_convert_s16(void *obj, int16_t *dst, unsigned int dst_frames, ++ const int16_t *src, unsigned int src_frames) ++{ ++ struct rate_src *rate = obj; ++ ++ do_convert(rate, dst, dst_frames, src, src_frames); + } + + static void pcm_src_close(void *obj) +@@ -167,12 +209,28 @@ static void dump(void *obj ATTRIBUTE_UNUSED, snd_output_t *out) + } + #endif + ++#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010003 ++static int get_supported_formats(void *obj, uint64_t *in_formats, ++ uint64_t *out_formats, ++ unsigned int *flags) ++{ ++ *in_formats = *out_formats = ++ (1ULL << SND_PCM_FORMAT_S16) | ++ (1ULL << SND_PCM_FORMAT_S32); ++ *flags = SND_PCM_RATE_FLAG_INTERLEAVED; ++ return 0; ++} ++#endif ++ + static snd_pcm_rate_ops_t pcm_src_ops = { + .close = pcm_src_close, + .init = pcm_src_init, + .free = pcm_src_free, + .reset = pcm_src_reset, + .adjust_pitch = pcm_src_adjust_pitch, ++#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010003 ++ .convert = pcm_src_convert, ++#endif + .convert_s16 = pcm_src_convert_s16, + .input_frames = input_frames, + .output_frames = output_frames, +@@ -181,6 +239,9 @@ static snd_pcm_rate_ops_t pcm_src_ops = { + .get_supported_rates = get_supported_rates, + .dump = dump, + #endif ++#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010003 ++ .get_supported_formats = get_supported_formats, ++#endif + }; + + static int pcm_src_open(unsigned int version, void **objp, +@@ -188,24 +249,27 @@ static int pcm_src_open(unsigned int version, void **objp, + { + struct rate_src *rate; + +-#if SND_PCM_RATE_PLUGIN_VERSION < 0x010002 +- if (version != SND_PCM_RATE_PLUGIN_VERSION) { +- fprintf(stderr, "Invalid rate plugin version %x\n", version); +- return -EINVAL; +- } +-#endif + rate = calloc(1, sizeof(*rate)); + if (! rate) + return -ENOMEM; ++ ++ rate->version = version; + rate->converter = type; + + *objp = rate; + #if SND_PCM_RATE_PLUGIN_VERSION >= 0x010002 +- if (version == 0x010001) ++ if (version == 0x010001) { + memcpy(ops, &pcm_src_ops, sizeof(snd_pcm_rate_old_ops_t)); +- else ++ return 0; ++ } ++#endif ++#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010003 ++ if (version == 0x010002) { ++ memcpy(ops, &pcm_src_ops, sizeof(snd_pcm_rate_v2_ops_t)); ++ return 0; ++ } + #endif +- *ops = pcm_src_ops; ++ *ops = pcm_src_ops; + return 0; + } + diff --git a/alsa-plugins-freeworld.spec b/alsa-plugins-freeworld.spec index 6f5404a..7810253 100644 --- a/alsa-plugins-freeworld.spec +++ b/alsa-plugins-freeworld.spec @@ -1,11 +1,12 @@ Name: alsa-plugins-freeworld Version: 1.2.5 -Release: 2%{?dist} +Release: 3%{?dist} Summary: The ALSA Plugins - freeworld version # All packages are LGPLv2+ with the exception of samplerate which is GPLv2+ License: LGPLv2+ URL: http://www.alsa-project.org/ Source0: https://www.alsa-project.org/files/pub/plugins/alsa-plugins-%{version}.ta... +Patch0: alsa-git.patch BuildRequires: autoconf automake libtool BuildRequires: alsa-lib-devel >= 1.1.8 @@ -40,6 +41,7 @@ The plugin uses ffmpeg audio resample library to convert audio rates. %prep %setup -q -n alsa-plugins-%{version}%{?prever} +%patch0 -p1 -b .alsa-git %build %configure --disable-static \ @@ -86,6 +88,9 @@ find %buildroot -name "*.la" -exec rm {} \; %changelog +* Tue Nov 9 2021 Jaroslav Kysela <perex(a)perex.cz&gt; - 1.2.5-3 +- Updated to latest ALSA git (use libswresample) + * Mon Aug 02 2021 RPM Fusion Release Engineering <leigh123linux(a)gmail.com&gt; - 1.2.5-2 - Rebuilt for https://fedoraproject.org/wiki/Fedora_35_Mass_Rebuild