Linux下使用ffmpeg播放mp3/aac/wav文件的音乐播放器应用
使用ffmpeg实现一个播放器？是不是没什么新意，不过一直使用ffmpeg程序，还没有用ffmpeg代码接口实现播放器，并且还需要使用linux的alsa接口播放出声音，所以做出来还是觉得有点意思；
需求：实现一个嵌入式linux上支持mp3/aac/wav文件的播放器
实现：所以考虑基于ffmpeg 实现一个嵌入式linux的播放器，这里主要应用ffmpeg的协议处理和音频解码能力，虽然网上的代码很多，不过由于版本的差异，例子程序接口存在差异，实现起来还是花了两天调试的时间；
0、几点总结：
---多看官方的例子程序，官方例子路径：\ffmpeg-4.1.9\tmp\share\ffmpeg\examples
---avcodec_open2失败，怎么处理？
关键函数：avcodec_parameters_to_context 将avcodec_find_decoder找到的音频解码器复制decoder；
---av_read_frame存在内存泄漏，怎么处理？
关键函数：av_packet_unref(&input_packet);
---alsa播放设备如何枚举？
关键函数：snd_device_name_get_hint
avcodec_receive_frame接收解码完的frame只用申请一次内存；
AVFrame *pframeSRC = av_frame_alloc();
这里ffmpeg使用版本：ffmpeg-4.1.9，编译选项：
//fdk-aacm
root@lyz-VirtualBox:/home/lyz/work/broadcast_app/app/thirds_libs_src/fdk-aac/build# vim arm-gcc-cxx11.cmake
root@lyz-VirtualBox:/home/lyz/work/broadcast_app/app/thirds_libs_src/fdk-aac/build# cmake -DCMAKE_TOOLCHAIN_FILE=/home/lyz/work/broadcast_app/app/thirds_libs_src/fdk-aac/build/arm-gcc-cxx11.cmake ../
-- The C compiler identification is GNU 6.4.1
root@lyz-VirtualBox:/home/lyz/work/broadcast_app/app_linux# cat /home/lyz/work/broadcast_app/app/thirds_libs_src/fdk-aac/build/arm-gcc-cxx11.cmake
# Sample toolchain file for building with gcc compiler
#
# Typical usage:
# *) cmake -H. -B_build -DCMAKE_TOOLCHAIN_FILE="${PWD}/toolchains/gcc.cmake"
SET(CMAKE_SYSTEM_NAME Linux)
set(CMAKE_SYSTEM_PROCESSOR arm)
# set compiler
set(CMAKE_C_COMPILER arm-openwrt-linux-gnueabi-gcc)
set(CMAKE_CXX_COMPILER arm-openwrt-linux-gnueabi-g++)
set(CONFIGURE_OPTS --enable-static=yes --enable-shared=no --disable-shared)
# set c++ standard
set(CMAKE_CXX_STANDARD 11)
set(CMAKE_CXX_STANDARD_REQUIRED ON)
set(CMAKE_CXX_EXTENSIONS OFF)
//mp3
/home/lyz/work/broadcast_app/app/thirds_libs_src/lame-3.100
./configure --host=arm-openwrt-linux-gnueabi --prefix=${PWD}/build/
./configure --target-os=linux --prefix=/home/lyz/work/broadcast_app/app_linux/thirds_libs_src/ffmpeg-4.1.9/tmp --disable-shared --disable-muxers --enable-pic --enable-static --enable-gpl --enable-nonfree --enable-ffmpeg --disable-debug --disable-filters --disable-encoders --disable-hwaccels --enable-static --enable-libmp3lame --enable-demuxers --enable-parsers --enable-protocols --disable-x86asm --disable-stripping --extra-cflags='-I/home/lyz/work/broadcast_app/app_linux/libs/include/ -I/home/lyz/work/broadcast_app/app_linux/libs/include/lame -Os -fpic ' --extra-ldflags='-ldl -lm -L/home/lyz/work/broadcast_app/app_linux/libs/' --enable-decoder=aac --enable-swresample --enable-decoder=ac3
1、cpp文件引用ffmpeg库，出现链接错误，需要在包括头文件的地方增加两个前缀：
//.cpp
#include <alsa/asoundlib.h>
#ifdef __cplusplus
extern "C" {
#endif
#include "libavutil/time.h"
#include "libavformat/avformat.h"
#include "libavcodec/avcodec.h"
#include "libavdevice/avdevice.h"
#include "libswresample/swresample.h"
#include "libswscale/swscale.h"
#ifdef __cplusplus
}
#endif2、上面修改后，还是出现链接错误，与链接库的链接顺序有关系；
错误的a库顺序：
LDFLAGS += -L ./libs/ -lavcodec -lavfilter -lavformat -lavutil -lpostproc -lswscale -lswresample -lfdk-aac -lmp3lame
正确的链接库顺序：
LDFLAGS += -Wl,-Bstatic -L./libs -lavformat -lavcodec -lswscale -lswresample -lavutil -lavfilter -lavdevice -lpostproc -lfdk-aac -lmp3lame
注意到动态链接和静态链接：
LDFLAGS += -Wl,-Bstatic -L./libs -lavformat -lavcodec -lswscale -lswresample -lavutil -lavfilter -lavdevice -lpostproc -lfdk-aac -lmp3lame
LDFLAGS += -Wl,-Bdynamic -ldl -lm -lasound -lpthread
3、内存泄漏，用valgrind 检查会有内存泄漏，播放一会就因为内存问题挂掉了；
使用valgrind可以很好的定位程序中的内存问题；
root@lyz-VirtualBox:/home/lyz/work/broadcast_app/app_linux# valgrind ./bas ./Test1.wav 0
4、使用alsa接口，完整播放出mp3文件声音的代码；
//static const char *device = "hw:1,0"; /* playback device "hw:0,0" */
static snd_pcm_format_t format = SND_PCM_FORMAT_S16; /* sample format */
static unsigned int rate = 44100; /* stream rate */
static unsigned int channels = 2; /* count of channels */
static unsigned int buffer_time = 500000; /* ring buffer length in us */
static unsigned int period_time = 100000; /* period time in us */
static int resample = 1; /* enable alsa-lib resampling */
static snd_pcm_sframes_t buffer_size;
static snd_pcm_sframes_t period_size;
snd_pcm_access_t mode = SND_PCM_ACCESS_RW_INTERLEAVED;
static snd_output_t *output = NULL;
/*配置参数*/
static int set_hwparams(snd_pcm_t *handle,snd_pcm_hw_params_t *params,snd_pcm_access_t access)
{
unsigned int rrate;
snd_pcm_uframes_t size;
int err, dir = 0;
/* choose all parameters */
err = snd_pcm_hw_params_any(handle, params);
if (err < 0) {
printf("Broken configuration for playback: no configurations available: %s\n", snd_strerror(err));
return err;
}
/* set hardware resampling */
err = snd_pcm_hw_params_set_rate_resample(handle, params, resample);
if (err < 0) {
printf("Resampling setup failed for playback: %s\n", snd_strerror(err));
return err;
}
/* set the interleaved read/write format */
/*访问格式*/
err = snd_pcm_hw_params_set_access(handle, params, mode);
if (err < 0) {
printf("Access type not available for playback: %s\n", snd_strerror(err));
return err;
}
/* set the sample format */
/*采样格式*/
err = snd_pcm_hw_params_set_format(handle, params, format);
if (err < 0) {
printf("Sample format not available for playback: %s\n", snd_strerror(err));
return err;
}
/* set the count of channels */
/*音频声道*/
err = snd_pcm_hw_params_set_channels(handle, params, channels);
if (err < 0) {
printf("Channels count (%u) not available for playbacks: %s\n", channels, snd_strerror(err));
return err;
}
/* set the stream rate */
/*采样率*/
rrate = rate;
err = snd_pcm_hw_params_set_rate_near(handle, params, &rrate, 0);
if (err < 0) {
printf("Rate %uHz not available for playback: %s\n", rate, snd_strerror(err));
return err;
}
if (rrate != rate) {
printf("Rate doesn't match (requested %uHz, get %iHz)\n", rate, err);
return -EINVAL;
}
/* set the buffer time */
/*底层buffer区间,以时间为单位，500000=0.5s*/
err = snd_pcm_hw_params_set_buffer_time_near(handle, params, &buffer_time, &dir);
if (err < 0) {
printf("Unable to set buffer time %u for playback: %s\n", buffer_time, snd_strerror(err));
return err;
}
err = snd_pcm_hw_params_get_buffer_size(params, &size);
if (err < 0) {
printf("Unable to get buffer size for playback: %s\n", snd_strerror(err));
return err;
}
buffer_size = size;
printf("buffer_size=%ld\n",buffer_size);
/* set the period time */
/*底层period区间,以时间为单位,100000=0.1s*/
err = snd_pcm_hw_params_set_period_time_near(handle, params, &period_time, &dir);
if (err < 0) {
printf("Unable to set period time %u for playback: %s\n", period_time, snd_strerror(err));
return err;
}
/*底层period区间，以字节为单位,44100*0.1=4410*/
err = snd_pcm_hw_params_get_period_size(params, &size, &dir);
if (err < 0) {
printf("Unable to get period size for playback: %s\n", snd_strerror(err));
return err;
}
period_size = size;
printf("period_size=%ld\n",period_size);
/* write the parameters to device */
err = snd_pcm_hw_params(handle, params);
if (err < 0) {
printf("Unable to set hw params for playback: %s\n", snd_strerror(err));
return err;
}
return 0;
}
/**
* Initialize one data packet for reading or writing.
* @param packet Packet to be initialized
*/
static void init_packet(AVPacket *packet)
{
av_init_packet(packet);
/* Set the packet data and size so that it is recognized as being empty. */
packet->data = NULL;
packet->size = 0;
}
int test_play_mp3(int argc, char *argv[])
{
int rc;
int size;
int got_picture;
int nb_data;
bool pkt_pending = false;
int audio_stream_idx;
char **hints, **n;
char *alsa_device_name;
if (argc < 2){
printf("please input filename!\r\n");
return 1;
}
printf("test_play_file filename :%s\r\n", argv[1]);
snd_pcm_t *handle;
snd_pcm_hw_params_t *hwparams;
snd_pcm_hw_params_alloca(&hwparams);
printf("Stream parameters are %uHz, %s, %u channels\n", rate, snd_pcm_format_name(format), channels);
int err;
/* Enumerate sound devices */
err = snd_device_name_hint(-1, "pcm", (void***)&hints);
if (err != 0){
printf("please snd_device_name_hint:%d\r\n", err);
return err;
}
#if 1
snd_lib_error_set_handler(alsa_error_handler);
#else
/* Set a null error handler prior to enumeration to suppress errors */
snd_lib_error_set_handler(null_alsa_error_handler);
#endif
n = hints;
while (*n != NULL) {
char *name = snd_device_name_get_hint(*n, "NAME");
if (name != NULL) {
if (0 != strcmp("null", name)){
snd_pcm_t* pcm;
int pb_result = snd_pcm_open (&pcm, name, SND_PCM_STREAM_PLAYBACK, 0);
if (pb_result >= 0) {
printf("Try to open the device for playback - success\r\n");
snd_pcm_close (pcm);
pcm = NULL;
alsa_device_name = name;
break;
}
printf("found device:%s\r\n", alsa_device_name);
//break;
}
}
n++;
}
printf("Playback device is %s\n", alsa_device_name);
/* Install error handler after enumeration, otherwise we'll get many
* error messages about invalid card/device ID.
*/
snd_lib_error_set_handler(alsa_error_handler);
err = snd_device_name_free_hint((void**)hints);
err = snd_output_stdio_attach(&output, stdout, 0);
if (err < 0) {
printf("Output failed: %s\n", snd_strerror(err));
return 0;
}
/*设置播放模式*/
err = snd_pcm_open(&handle, alsa_device_name, SND_PCM_STREAM_PLAYBACK, 0);
if (err < 0)
{
printf("Playback open error: %s\n", snd_strerror(err));
return 0;
}
/*设置参数*/
err = set_hwparams(handle, hwparams, mode);
if (err < 0) {
printf("Setting of hwparams failed: %s\n", snd_strerror(err));
return 0;
}
//period_size大概是采样点数/帧——4410点/帧
//s16位代表两个字节，再加上双声道
//size公式=period_size*channels*16/8
size = (period_size * channels * snd_pcm_format_physical_width(format)) / 8; /* 2 bytes/sample, 1 channels */
printf("size:%d\n",size);
char *buffer;
buffer = (char *) malloc(size);
memset(buffer,0,size);
char *in_name=argv[1];//"鄧紫棋 - 睡公主.wav";
int ret;
AVFormatContext* infmt_ctx = NULL;
//创建输入封装器
ret=avformat_open_input(&infmt_ctx, in_name, NULL, NULL);
if (ret != 0)
{
printf("failed alloc output context\n");
return -1;
}
infmt_ctx->max_analyze_duration = 5*AV_TIME_BASE;
//读取一部分视音频流并且获得一些相关的信息
ret=avformat_find_stream_info(infmt_ctx, NULL);
if (ret < 0) {
printf("Can't get stream info\n");
avformat_close_input(&infmt_ctx);
return -1;
}
audio_stream_idx = av_find_best_stream(infmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, NULL, 0);
if (audio_stream_idx < 0) {
printf( "Can't find video stream in input file\n");
avformat_close_input(&infmt_ctx);
return -1;
}
AVCodecParameters *pCodecParameters = infmt_ctx->streams[audio_stream_idx]->codecpar;
if (pCodecParameters == NULL){
printf("pCodecParameters is NULL\n");
avformat_close_input(&infmt_ctx);
return -1;
}
//找到解码器
const AVCodec* decodec = avcodec_find_decoder(pCodecParameters->codec_id);
if (!decodec)
{
printf("not find decoder codec audio_stream_idx:%d codec_id:%d\n", audio_stream_idx, pCodecParameters->codec_id);
avformat_close_input(&infmt_ctx);
return -1;
}
AVCodecContext *decodec_ctx = avcodec_alloc_context3(decodec);
if (!decodec_ctx) {
printf("Can't allocate decoder context\n");
avformat_close_input(&infmt_ctx);
return AVERROR(ENOMEM);
}
if(avcodec_parameters_to_context(decodec_ctx, pCodecParameters)<0){
printf("Cannot alloc codec context.\n");
avformat_close_input(&infmt_ctx);
return -1;
}
decodec_ctx->pkt_timebase = infmt_ctx->streams[audio_stream_idx]->time_base;
#if 0
decodec_ctx->sample_rate = pCodecParameters->sample_rate;
decodec_ctx->sample_fmt = (AVSampleFormat)pCodecParameters->format ;
decodec_ctx->channels = pCodecParameters->channels;
decodec_ctx->channel_layout = pCodecParameters->channel_layout;
#endif//
//打开解码器
ret = avcodec_open2(decodec_ctx, decodec, NULL);
if (ret < 0) {
printf("Could not open codec: %d\n", ret);
avformat_close_input(&infmt_ctx);
return -1;
}
//查看输入封装内容
av_dump_format(infmt_ctx, 0, in_name,0);
#if 1
AVFrame *pframePCM = av_frame_alloc();
pframePCM->format = AV_SAMPLE_FMT_S16;
pframePCM->channel_layout = AV_CH_LAYOUT_STEREO;
pframePCM->sample_rate = rate;
pframePCM->nb_samples = period_size;
pframePCM->channels = channels;
av_frame_get_buffer(pframePCM, 0);
#else
uint8_t *converted_input_samples = NULL;
int converted_input_samples_size = av_samples_alloc(&converted_input_samples, NULL,
channels ,
period_size,
AV_SAMPLE_FMT_S16, 0);
#endif
struct SwrContext *pcm_convert_ctx = swr_alloc();
if (!pcm_convert_ctx)
{
printf("Could not allocate resampler context\n");
free(buffer);
return -1;
}
swr_alloc_set_opts(pcm_convert_ctx,
AV_CH_LAYOUT_STEREO,
AV_SAMPLE_FMT_S16,
pframePCM->sample_rate,
av_get_default_channel_layout(decodec_ctx->channels),
decodec_ctx->sample_fmt,
decodec_ctx->sample_rate,
0,
NULL);
ret = swr_init(pcm_convert_ctx);
if (ret<0)
{
printf("Failed to initialize the resampling context\n");
free(buffer);
return -1;
}
AVPacket *input_packet=av_packet_alloc();
init_packet(input_packet);
AVFrame *pframeSRC = av_frame_alloc();
#if 0
pframeSRC->format = (AVSampleFormat)pCodecParameters->format ;
pframeSRC->channel_layout = decodec_ctx->channel_layout;
pframeSRC->sample_rate = decodec_ctx->sample_rate;
pframeSRC->nb_samples = (20*decodec_ctx->sample_rate * channels * 2) / 8000;;
pframeSRC->channels = channels;
av_frame_get_buffer(pframeSRC, 0);
#endif
int finished = 0;
int decode_ret = 0;
int data_size = av_get_bytes_per_sample(decodec_ctx->sample_fmt);
printf("data_size:%d, frame_size:%d, dst_samples:%d\n", data_size, pCodecParameters->frame_size, pframePCM->nb_samples);
while (!finished)
{
ret=av_read_frame(infmt_ctx, input_packet);
if (ret != 0)
{
if (ret == AVERROR_EOF){
finished = 1;
break;
}
printf("fail to read_frame\n");
break;
}
//avcodec_send_packet/avcodec_receive_frame
//针对多音轨问题处理
if(input_packet->stream_index != audio_stream_idx){
av_packet_unref(input_packet);
continue;
}
//解码获取初始音频
ret = avcodec_send_packet(decodec_ctx, input_packet);
if (ret == AVERROR(EAGAIN)) {
pkt_pending = true;
continue;
}if (ret < 0){
break;
}
int no_resample = 0;
do{
decode_ret = avcodec_receive_frame(decodec_ctx, pframeSRC);
if (decode_ret == AVERROR_EOF) {//取完数据帧复位解码器
//avcodec_flush_buffers(decodec_ctx);
printf("avcodec_receive_frame eof\n");
//av_packet_unref(&input_packet);
break;
} else if (decode_ret < 0){
break;
}
int source_samples = swr_get_out_samples(pcm_convert_ctx, pframeSRC->nb_samples);
int out_samples = source_samples;//
uint8_t *write_2_pcm = NULL;
if (out_samples != pframePCM->nb_samples){
no_resample = 1;
//读取到一帧音频或者视频
//MP3->PCM，
ret=swr_convert(pcm_convert_ctx, pframePCM->data, pframePCM->nb_samples,(const uint8_t **)pframeSRC->extended_data, pframeSRC->nb_samples);
if (ret <= 0)
{
printf("[0]out_samples:%d, pframeSRC->nb_samples:%d,ret:%d\n", source_samples, pframeSRC->nb_samples, ret);
continue;
}else{
//printf("[2]out_samples:%d, pframeSRC->nb_samples:%d,ret:%d\n", source_samples, pframeSRC->nb_samples, ret);
}
write_2_pcm = pframePCM->data[0];
nb_data = ret;
}else{
printf("out_samples:%d, pframeSRC->nb_samples:%d \n", out_samples, pframeSRC->nb_samples );
nb_data = out_samples;
write_2_pcm = pframeSRC->data[0];
}
//向硬件写入音频数据
rc = snd_pcm_writei(handle, write_2_pcm, out_samples);
if (rc == -EPIPE) {
printf("underrun occurred\n");
err=snd_pcm_prepare(handle);
if(err<0)
{
printf("can not recover from underrun: %s\n",snd_strerror(err));
}
}
else if (rc < 0) {
fprintf(stderr,"error from writei: %s\n",snd_strerror(rc));
}
else if (rc != (int)nb_data) {
fprintf(stderr,"short write, write %d frames\n", rc);
}
}while(no_resample && decode_ret > 0);
av_packet_unref(input_packet);
}
if (pcm_convert_ctx) {
swr_free(&pcm_convert_ctx);
}
av_packet_free(&input_packet);
if (pframeSRC) {
av_frame_free(&pframeSRC);
}
#if 1
if (pframePCM) {
av_frame_free(&pframePCM);
}
#endif
if(decodec_ctx != NULL){
avcodec_close(decodec_ctx);
avcodec_free_context(&decodec_ctx);
}
if (infmt_ctx != NULL) {
avformat_close_input(&infmt_ctx);
avformat_free_context(infmt_ctx);
}
snd_pcm_drain(handle);
snd_pcm_close(handle);
//free(converted_input_samples);
free(buffer);
free(alsa_device_name);
return 0;
}参考：https://blog.csdn.net/pk296256948/article/details/113695358
下一步：实现对rtsp流的请求；
--
2022/11/28更新：实现rtsp播放器，只需要将播放路径直接给一个rtsp的地址就可以了，是不是很简单！
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