ffmpeg pcm混音
视频会议中经常需要处理的场景有多路音频混音，那么混音有很多种算法有比较主流的有归一权重、叠加均值、平均权重等方法；如果公司要开发生产级别的音频混合要的算法可能会更加多，可以找算法公司购买。
ffmpeg也有混音的操作，用的就是平均权重算法。
#define ENABLE_FILTERS 1
static const char* filter_descr = "[in0][in1]amix=inputs=2[out]";//"aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
static const char* player = "ffplay -f s16le -ar 8000 -ac 1 -";
static AVFormatContext* fmt_ctx1;
static AVFormatContext* fmt_ctx2;
static AVCodecContext* dec_ctx1;
static AVCodecContext* dec_ctx2;
AVFilterContext* buffersink_ctx;
AVFilterContext* buffersrc_ctx1;
AVFilterContext* buffersrc_ctx2;
AVFilterGraph* filter_graph;
static int audio_stream_index_1 = -1;
static int audio_stream_index_2 = -1;
static int open_input_file_1(const char* filename)
{
int ret;
AVCodec* dec;
if ((ret = avformat_open_input(&fmt_ctx1, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(fmt_ctx1, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
/* select the audio stream */
ret = av_find_best_stream(fmt_ctx1, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
if (ret streams[audio_stream_index_1]->codec;
av_opt_set_int(dec_ctx1, "refcounted_frames", 1, 0);
/* init the audio decoder */
if ((ret = avcodec_open2(dec_ctx1, dec, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
return ret;
}
return 0;
}
static int open_input_file_2(const char* filename)
{
int ret;
AVCodec* dec;
if ((ret = avformat_open_input(&fmt_ctx2, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(fmt_ctx2, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
/* select the audio stream */
ret = av_find_best_stream(fmt_ctx2, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
if (ret streams[audio_stream_index_2]->codec;
av_opt_set_int(dec_ctx2, "refcounted_frames", 1, 0);
/* init the audio decoder */
if ((ret = avcodec_open2(dec_ctx2, dec, NULL)) streams[audio_stream_index_1]->time_base;
AVRational time_base_2 = fmt_ctx2->streams[audio_stream_index_2]->time_base;
filter_graph = avfilter_graph_alloc();
if (!outputs1 || !inputs || !filter_graph) {
ret = AVERROR(ENOMEM);
goto end;
}
/* buffer audio source: the decoded frames from the decoder will be inserted here. */
if (!dec_ctx1->channel_layout)
dec_ctx1->channel_layout = av_get_default_channel_layout(dec_ctx1->channels);
snprintf(args1, sizeof(args1),
"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
time_base_1.num, time_base_1.den, dec_ctx1->sample_rate,
av_get_sample_fmt_name(dec_ctx1->sample_fmt), dec_ctx1->channel_layout);
ret = avfilter_graph_create_filter(&buffersrc_ctx1, abuffersrc1, "in1",
args1, NULL, filter_graph);
if (ret channel_layout)
dec_ctx2->channel_layout = av_get_default_channel_layout(dec_ctx2->channels);
snprintf(args2, sizeof(args2),
"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
time_base_2.num, time_base_2.den, dec_ctx2->sample_rate,
av_get_sample_fmt_name(dec_ctx2->sample_fmt), dec_ctx2->channel_layout);
ret = avfilter_graph_create_filter(&buffersrc_ctx2, abuffersrc1, "in2",
args2, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
goto end;
}
#endif
/* buffer audio sink: to terminate the filter chain. */
ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
goto end;
}
ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
goto end;
}
ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
goto end;
}
ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret name = av_strdup("in0");
outputs1->filter_ctx = buffersrc_ctx1;
outputs1->pad_idx = 0;
#if (ENABLE_FILTERS)
outputs1->next = outputs2;
outputs2->name = av_strdup("in1");
outputs2->filter_ctx = buffersrc_ctx2;
outputs2->pad_idx = 0;
outputs2->next = NULL;
#else
outputs1->next = NULL;
#endif
/*
* The buffer sink input must be connected to the output pad of
* the last filter described by filters_descr; since the last
* filter output label is not specified, it is set to "out" by
* default.
*/
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
AVFilterInOut* filter_outputs[2];
filter_outputs[0] = outputs1;
#if (ENABLE_FILTERS)
filter_outputs[1] = outputs2;
#endif
if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
&inputs, &outputs1, NULL)) < 0)//filter_outputs
{
av_log(NULL, AV_LOG_ERROR, "parse ptr fail, ret: %d\n", ret);
goto end;
}
if ((ret = avfilter_graph_config(filter_graph, NULL)) inputs[0];
av_get_channel_layout_string(args1, sizeof(args1), -1, outlink->channel_layout);
av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
(int)outlink->sample_rate,
(char*)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
args1);
end:
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs1);
return ret;
}
static void print_frame(const AVFrame* frame)
#if 1
{
FILE* file = NULL;
const int n = frame->nb_samples * av_get_channel_layout_nb_channels(av_frame_get_channel_layout(frame));
const uint16_t* p = (uint16_t*)frame->data[0];
const uint16_t* p_end = p + n;
file = fopen("C:/Users/liyihang/Desktop/zhuo_main/sc/tmp4.pcm", "ab+");
if (NULL == file) {
perror("fopen tmp.mp3 error\n");
return;
}
else {
perror("fopen tmp.aac successful\n");
}
fwrite(frame->data[0], n * 2, 1, file);
fclose(file);
file = NULL;
}
#else
{
const int n = frame->nb_samples * av_get_channel_layout_nb_channels(av_frame_get_channel_layout(frame));
const uint16_t* p = (uint16_t*)frame->data[0];
const uint16_t* p_end = p + n;
while (p > 8 & 0xff, stdout);
p++;
}
fflush(stdout);
}
#endif
int main2(int argc, char** argv)
{
int ret;
AVFrame* frame = av_frame_alloc();
AVFrame* filt_frame = av_frame_alloc();
int got_frame;
if (!frame || !filt_frame) {
perror("Could not allocate frame");
exit(1);
}
/*
if (argc != 2) {
fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
exit(1);
}
*/
av_register_all();
avfilter_register_all();
if ((ret = open_input_file_1("C:/Users/liyihang/Desktop/zhuo_main/sc/bs.mp4")) < 0)
{
av_log(NULL, AV_LOG_ERROR, "open input file fail, ret: %d\n", ret);
goto end;
}
if ((ret = open_input_file_2("C:/Users/liyihang/Desktop/zhuo_main/sc/bs2.mp4")) < 0)
{
av_log(NULL, AV_LOG_ERROR, "open input file fail, ret: %d\n", ret);
goto end;
}
if ((ret = init_filters(filter_descr)) < 0)
{
av_log(NULL, AV_LOG_ERROR, "init filters fail, ret: %d\n", ret);
goto end;
}
AVPacket packet0, packet;
AVPacket _packet0, _packet;
/* read all packets */
packet0.data = NULL;
packet.data = NULL;
_packet0.data = NULL;
_packet.data = NULL;
while (1) {
if (!packet0.data) {
if ((ret = av_read_frame(fmt_ctx1, &packet)) < 0)
break;
packet0 = packet;
}
if (packet.stream_index == audio_stream_index_1) {
got_frame = 0;
ret = avcodec_decode_audio4(dec_ctx1, frame, &got_frame, &packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
continue;
}
packet.size -= ret;
packet.data += ret;
if (got_frame) {
av_log(NULL, AV_LOG_ERROR, "push frame\n");
/* push the audio data from decoded frame into the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx1, frame, 0) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
break;
}
av_log(NULL, AV_LOG_ERROR, "pull frame\n");
}
if (packet.size <= 0)
av_packet_unref(&packet0);
}
else {
/* discard non-wanted packets */
av_packet_unref(&packet0);
}
if (!_packet0.data) {
if ((ret = av_read_frame(fmt_ctx2, &_packet)) < 0)
break;
_packet0 = _packet;
}
if (_packet.stream_index == audio_stream_index_2) {
got_frame = 0;
ret = avcodec_decode_audio4(dec_ctx2, frame, &got_frame, &_packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
continue;
}
_packet.size -= ret;
_packet.data += ret;
if (got_frame) {
av_log(NULL, AV_LOG_ERROR, "push frame\n");
/* push the audio data from decoded frame into the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx2, frame, 0) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
break;
}
av_log(NULL, AV_LOG_ERROR, "pull frame\n");
}
if (_packet.size <= 0)
av_packet_unref(&_packet0);
}
else {
/* discard non-wanted packets */
av_packet_unref(&_packet0);
}
/* pull filtered audio from the filtergraph */
if (got_frame)
{
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if (ret < 0)
{
av_log(NULL, AV_LOG_ERROR, "buffersink get frame fail, ret: %d\n", ret);
goto end;
}
print_frame(filt_frame);
av_frame_unref(filt_frame);
}
}
}
end:
avfilter_graph_free(&filter_graph);
avcodec_close(dec_ctx1);
avformat_close_input(&fmt_ctx1);
avcodec_close(dec_ctx2);
avformat_close_input(&fmt_ctx2);
av_frame_free(&frame);
av_frame_free(&filt_frame);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
exit(1);
}
exit(0);
}
使用的ffmpeg filter机制完成混音。
作者：一航jason
