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-rw-r--r--src/main.cpp172
1 files changed, 89 insertions, 83 deletions
diff --git a/src/main.cpp b/src/main.cpp
index 7de6cf0..c293338 100644
--- a/src/main.cpp
+++ b/src/main.cpp
@@ -565,44 +565,42 @@ static void int_handler(int dummy) {
running = 0;
}
+struct Arg {
+ const char *value;
+ bool optional;
+};
+
int main(int argc, char **argv) {
signal(SIGINT, int_handler);
- std::map<std::string, std::string> args = {
- { "-w", "" },
- { "-c", "" },
- { "-f", "" },
- { "-a", "" },
+ std::map<std::string, Arg> args = {
+ { "-w", Arg { nullptr, false } },
+ { "-c", Arg { nullptr, false } },
+ { "-f", Arg { nullptr, false } },
+ { "-a", Arg { nullptr, true } }
};
- for(int i = 1; i < argc; i += 2) {
- bool valid_arg = false;
- for(auto &it : args) {
- if(strcmp(argv[i], it.first.c_str()) == 0) {
- it.second = argv[i + 1];
- valid_arg = true;
- break;
- }
- }
-
- if(!valid_arg) {
+ for(int i = 1; i < argc - 1; i += 2) {
+ auto it = args.find(argv[i]);
+ if(it == args.end()) {
fprintf(stderr, "Invalid argument '%s'\n", argv[i]);
usage();
}
+ it->second.value = argv[i + 1];
}
for(auto &it : args) {
- if(it.second.empty()) {
+ if(!it.second.optional && !it.second.value) {
fprintf(stderr, "Missing argument '%s'\n", it.first.c_str());
usage();
}
}
- Window src_window_id = strtol(args["-w"].c_str(), nullptr, 0);
- const char *container_format = args["-c"].c_str();
- int fps = atoi(args["-f"].c_str());
+ Window src_window_id = strtol(args["-w"].value, nullptr, 0);
+ const char *container_format = args["-c"].value;
+ int fps = atoi(args["-f"].value);
if(fps <= 0 || fps > 255) {
- fprintf(stderr, "invalid fps argument: %s\n", args["-f"].c_str());
+ fprintf(stderr, "invalid fps argument: %s\n", args["-f"].value);
return 1;
}
@@ -807,67 +805,74 @@ int main(int argc, char **argv) {
int window_width = xwa.width;
int window_height = xwa.height;
+ std::mutex write_output_mutex;
+ std::thread audio_thread;
+
SoundDevice sound_device;
- if(sound_device_get_by_name(&sound_device, args["-a"].c_str(), audio_stream->codec->channels, audio_stream->codec->frame_size) != 0) {
- fprintf(stderr, "failed to get 'pulse' sound device\n");
- exit(1);
- }
+ Arg &audio_input_arg = args["-a"];
+ if(audio_input_arg.value) {
+ if(sound_device_get_by_name(&sound_device, audio_input_arg.value, audio_stream->codec->channels, audio_stream->codec->frame_size) != 0) {
+ fprintf(stderr, "failed to get 'pulse' sound device\n");
+ exit(1);
+ }
+
+ int audio_buffer_size = av_samples_get_buffer_size(NULL, audio_stream->codec->channels, audio_stream->codec->frame_size, audio_stream->codec->sample_fmt, 1);
+ uint8_t *audio_frame_buf = (uint8_t *)av_malloc(audio_buffer_size);
+ avcodec_fill_audio_frame(audio_frame, audio_stream->codec->channels, audio_stream->codec->sample_fmt, (const uint8_t*)audio_frame_buf, audio_buffer_size, 1);
- int audio_buffer_size = av_samples_get_buffer_size(NULL, audio_stream->codec->channels, audio_stream->codec->frame_size, audio_stream->codec->sample_fmt, 1);
- uint8_t *audio_frame_buf = (uint8_t *)av_malloc(audio_buffer_size);
- avcodec_fill_audio_frame(audio_frame, audio_stream->codec->channels, audio_stream->codec->sample_fmt, (const uint8_t*)audio_frame_buf, audio_buffer_size, 1);
-
- AVPacket audio_packet;
- av_new_packet(&audio_packet, audio_buffer_size);
-
- std::mutex write_output_mutex;
-
- std::thread audio_thread([](AVFormatContext *av_format_context, AVStream *audio_stream, AVPacket *audio_packet, uint8_t *audio_frame_buf, SoundDevice *sound_device, AVFrame *audio_frame, std::mutex *write_output_mutex) mutable {
- SwrContext *swr = swr_alloc();
- if(!swr) {
- fprintf(stderr, "Failed to create SwrContext\n");
- exit(1);
- }
- av_opt_set_int(swr, "in_channel_layout", audio_stream->codec->channel_layout, 0);
- av_opt_set_int(swr, "out_channel_layout", audio_stream->codec->channel_layout, 0);
- av_opt_set_int(swr, "in_sample_rate", audio_stream->codec->sample_rate, 0);
- av_opt_set_int(swr, "out_sample_rate", audio_stream->codec->sample_rate, 0);
- av_opt_set_sample_fmt(swr, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
- av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
- swr_init(swr);
-
- while(running) {
- void *sound_buffer;
- int sound_buffer_size = sound_device_read_next_chunk(sound_device, &sound_buffer);
- if(sound_buffer_size >= 0) {
- // TODO: Instead of converting audio, get float audio from alsa. Or does alsa do conversion internally to get this format?
- swr_convert(swr, &audio_frame_buf, audio_frame->nb_samples, (const uint8_t**)&sound_buffer, sound_buffer_size);
- audio_frame->extended_data = &audio_frame_buf;
- // TODO: Fix this. Warning from ffmpeg:
- // Timestamps are unset in a packet for stream 1. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly
- //audio_frame->pts=audio_frame_index*100;
- //++audio_frame_index;
-
- int got_frame = 0;
- int ret = avcodec_encode_audio2(audio_stream->codec, audio_packet, audio_frame, &got_frame);
- if(ret < 0){
- printf("Failed to encode!\n");
- break;
- }
- if (got_frame==1){
- //printf("Succeed to encode 1 frame! \tsize:%5d\n",pkt.size);
- audio_packet->stream_index = audio_stream->index;
- std::lock_guard<std::mutex> lock(*write_output_mutex);
- ret = av_write_frame(av_format_context, audio_packet);
- av_free_packet(audio_packet);
- }
- } else {
- fprintf(stderr, "failed to read sound from device, error: %d\n", sound_buffer_size);
- }
- }
-
- swr_free(&swr);
- }, av_format_context, audio_stream, &audio_packet, audio_frame_buf, &sound_device, audio_frame, &write_output_mutex);
+ audio_thread = std::thread([audio_buffer_size](AVFormatContext *av_format_context, AVStream *audio_stream, uint8_t *audio_frame_buf, SoundDevice *sound_device, AVFrame *audio_frame, std::mutex *write_output_mutex) mutable {
+ AVPacket audio_packet;
+ if(av_new_packet(&audio_packet, audio_buffer_size) != 0) {
+ fprintf(stderr, "Failed to create audio packet\n");
+ exit(1);
+ }
+
+ SwrContext *swr = swr_alloc();
+ if(!swr) {
+ fprintf(stderr, "Failed to create SwrContext\n");
+ exit(1);
+ }
+ av_opt_set_int(swr, "in_channel_layout", audio_stream->codec->channel_layout, 0);
+ av_opt_set_int(swr, "out_channel_layout", audio_stream->codec->channel_layout, 0);
+ av_opt_set_int(swr, "in_sample_rate", audio_stream->codec->sample_rate, 0);
+ av_opt_set_int(swr, "out_sample_rate", audio_stream->codec->sample_rate, 0);
+ av_opt_set_sample_fmt(swr, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
+ av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
+ swr_init(swr);
+
+ while(running) {
+ void *sound_buffer;
+ int sound_buffer_size = sound_device_read_next_chunk(sound_device, &sound_buffer);
+ if(sound_buffer_size >= 0) {
+ // TODO: Instead of converting audio, get float audio from alsa. Or does alsa do conversion internally to get this format?
+ swr_convert(swr, &audio_frame_buf, audio_frame->nb_samples, (const uint8_t**)&sound_buffer, sound_buffer_size);
+ audio_frame->extended_data = &audio_frame_buf;
+ // TODO: Fix this. Warning from ffmpeg:
+ // Timestamps are unset in a packet for stream 1. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly
+ //audio_frame->pts=audio_frame_index*100;
+ //++audio_frame_index;
+
+ int got_frame = 0;
+ int ret = avcodec_encode_audio2(audio_stream->codec, &audio_packet, audio_frame, &got_frame);
+ if(ret < 0){
+ printf("Failed to encode!\n");
+ break;
+ }
+ if (got_frame==1){
+ //printf("Succeed to encode 1 frame! \tsize:%5d\n",pkt.size);
+ audio_packet.stream_index = audio_stream->index;
+ std::lock_guard<std::mutex> lock(*write_output_mutex);
+ ret = av_write_frame(av_format_context, &audio_packet);
+ av_free_packet(&audio_packet);
+ }
+ } else {
+ fprintf(stderr, "failed to read sound from device, error: %d\n", sound_buffer_size);
+ }
+ }
+
+ swr_free(&swr);
+ }, av_format_context, audio_stream, audio_frame_buf, &sound_device, audio_frame, &write_output_mutex);
+ }
bool redraw = true;
XEvent e;
@@ -1000,9 +1005,10 @@ int main(int argc, char **argv) {
}
running = 0;
- audio_thread.join();
-
- sound_device_close(&sound_device);
+ if(audio_input_arg.value) {
+ audio_thread.join();
+ sound_device_close(&sound_device);
+ }
//Flush Encoder
#if 0
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-06 13:26:43 +0000