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change to new api

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This commit is contained in:
William Bell
2025-12-27 21:33:02 +00:00
parent 65a5a1ee15
commit 74b8054ea2
3 changed files with 160 additions and 175 deletions
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2
.gitignore vendored
View File

@@ -177,3 +177,5 @@ cython_debug/
music
logs
data
bin

2
Makefile
View File

@@ -1,5 +1,5 @@
CC := gcc
TARGET := player
TARGET := bin/player
SRC_DIR := src
BUILD := build

91
src/main.c
View File

@@ -1,13 +1,13 @@
// player.c
#include <errno.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <errno.h>
#include <libavformat/avformat.h>
#include <libavcodec/avcodec.h>
#include <libswresample/swresample.h>
#include <libavformat/avformat.h>
#include <libavutil/opt.h>
#include <libswresample/swresample.h>
#include <alsa/asoundlib.h>
@@ -56,8 +56,7 @@ int main(int argc, char **argv) {
/* ---------- Decoder ---------- */
AVCodec *codec =
avcodec_find_decoder(stream->codecpar->codec_id);
const AVCodec *codec = avcodec_find_decoder(stream->codecpar->codec_id);
if (!codec) {
fprintf(stderr, "decoder not found\n");
return 1;
@@ -73,18 +72,27 @@ int main(int argc, char **argv) {
/* ---------- Resampler ---------- */
SwrContext *swr = swr_alloc_set_opts(
NULL,
OUT_LAYOUT,
OUT_FORMAT,
OUT_RATE,
dec->channel_layout ? dec->channel_layout
: av_get_default_channel_layout(dec->channels),
dec->sample_fmt,
dec->sample_rate,
0,
NULL
);
SwrContext *swr = swr_alloc();
if (!swr) {
fprintf(stderr, "failed to alloc swr\n");
return 1;
}
AVChannelLayout out_ch_layout;
av_channel_layout_default(&out_ch_layout, OUT_CHANNELS);
av_opt_set_chlayout(swr, "out_chlayout", &out_ch_layout, 0);
av_opt_set_int(swr, "out_sample_rate", OUT_RATE, 0);
av_opt_set_sample_fmt(swr, "out_sample_fmt", OUT_FORMAT, 0);
av_opt_set_chlayout(swr, "in_chlayout", &dec->ch_layout, 0);
av_opt_set_int(swr, "in_sample_rate", dec->sample_rate, 0);
av_opt_set_sample_fmt(swr, "in_sample_fmt", dec->sample_fmt, 0);
if (swr_init(swr) < 0) {
fprintf(stderr, "failed to init swr\n");
return 1;
}
if (!swr || swr_init(swr) < 0) {
fprintf(stderr, "failed to init resampler\n");
@@ -94,19 +102,13 @@ int main(int argc, char **argv) {
/* ---------- ALSA ---------- */
snd_pcm_t *pcm;
if (snd_pcm_open(&pcm, "default",
SND_PCM_STREAM_PLAYBACK, 0) < 0) {
if (snd_pcm_open(&pcm, "default", SND_PCM_STREAM_PLAYBACK, 0) < 0) {
fprintf(stderr, "failed to open ALSA device\n");
return 1;
}
snd_pcm_set_params(
pcm,
SND_PCM_FORMAT_S16_LE,
SND_PCM_ACCESS_RW_INTERLEAVED,
OUT_CHANNELS,
OUT_RATE,
1,
snd_pcm_set_params(pcm, SND_PCM_FORMAT_S16_LE, SND_PCM_ACCESS_RW_INTERLEAVED,
OUT_CHANNELS, OUT_RATE, 1,
500000 // 0.5s latency
);
@@ -130,33 +132,19 @@ int main(int argc, char **argv) {
avcodec_send_packet(dec, pkt);
while (avcodec_receive_frame(dec, frm) == 0) {
int needed = av_rescale_rnd(
swr_get_delay(swr, dec->sample_rate) + frm->nb_samples,
OUT_RATE,
dec->sample_rate,
AV_ROUND_UP
);
int needed =
av_rescale_rnd(swr_get_delay(swr, dec->sample_rate) + frm->nb_samples,
OUT_RATE, dec->sample_rate, AV_ROUND_UP);
if (needed > max_samples) {
av_freep(&outbuf);
av_samples_alloc(
&outbuf,
&out_linesize,
OUT_CHANNELS,
needed,
OUT_FORMAT,
1
);
av_samples_alloc(&outbuf, &out_linesize, OUT_CHANNELS, needed,
OUT_FORMAT, 1);
max_samples = needed;
}
int samples = swr_convert(
swr,
&outbuf,
needed,
(const uint8_t **)frm->data,
frm->nb_samples
);
int samples = swr_convert(swr, &outbuf, needed,
(const uint8_t **)frm->data, frm->nb_samples);
int written = snd_pcm_writei(pcm, outbuf, samples);
if (written < 0)
@@ -170,13 +158,8 @@ int main(int argc, char **argv) {
avcodec_send_packet(dec, NULL);
while (avcodec_receive_frame(dec, frm) == 0) {
int samples = swr_convert(
swr,
&outbuf,
max_samples,
(const uint8_t **)frm->data,
frm->nb_samples
);
int samples = swr_convert(swr, &outbuf, max_samples,
(const uint8_t **)frm->data, frm->nb_samples);
snd_pcm_writei(pcm, outbuf, samples);
}