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-rw-r--r--src/main.cpp49
1 files changed, 26 insertions, 23 deletions
diff --git a/src/main.cpp b/src/main.cpp
index e7aa115..71452e6 100644
--- a/src/main.cpp
+++ b/src/main.cpp
@@ -549,7 +549,7 @@ static const AVCodec* find_av1_encoder(gsr_gpu_vendor vendor, const char *card_p
return checked_success ? codec : nullptr;
}
-static AVFrame* open_audio(AVCodecContext *audio_codec_context) {
+static void open_audio(AVCodecContext *audio_codec_context) {
AVDictionary *options = nullptr;
av_dict_set(&options, "strict", "experimental", 0);
@@ -559,7 +559,9 @@ static AVFrame* open_audio(AVCodecContext *audio_codec_context) {
fprintf(stderr, "failed to open codec, reason: %s\n", av_error_to_string(ret));
_exit(1);
}
+}
+static AVFrame* create_audio_frame(AVCodecContext *audio_codec_context) {
AVFrame *frame = av_frame_alloc();
if(!frame) {
fprintf(stderr, "failed to allocate audio frame\n");
@@ -576,7 +578,7 @@ static AVFrame* open_audio(AVCodecContext *audio_codec_context) {
av_channel_layout_copy(&frame->ch_layout, &audio_codec_context->ch_layout);
#endif
- ret = av_frame_get_buffer(frame, 0);
+ int ret = av_frame_get_buffer(frame, 0);
if(ret < 0) {
fprintf(stderr, "failed to allocate audio data buffers, reason: %s\n", av_error_to_string(ret));
_exit(1);
@@ -924,13 +926,13 @@ struct AudioDevice {
SoundDevice sound_device;
AudioInput audio_input;
AVFilterContext *src_filter_ctx = nullptr;
+ AVFrame *frame = nullptr;
std::thread thread; // TODO: Instead of having a thread for each track, have one thread for all threads and read the data with non-blocking read
};
// TODO: Cleanup
struct AudioTrack {
AVCodecContext *codec_context = nullptr;
- AVFrame *frame = nullptr;
AVStream *stream = nullptr;
std::vector<AudioDevice> audio_devices;
@@ -1980,7 +1982,7 @@ int main(int argc, char **argv) {
if(replay_buffer_size_secs == -1)
audio_stream = create_stream(av_format_context, audio_codec_context);
- AVFrame *audio_frame = open_audio(audio_codec_context);
+ open_audio(audio_codec_context);
if(audio_stream)
avcodec_parameters_from_context(audio_stream->codecpar, audio_codec_context);
@@ -2020,11 +2022,13 @@ int main(int argc, char **argv) {
if(audio_input.name.empty()) {
audio_device.sound_device.handle = NULL;
audio_device.sound_device.frames = 0;
+ audio_device.frame = NULL;
} else {
if(sound_device_get_by_name(&audio_device.sound_device, audio_input.name.c_str(), audio_input.description.c_str(), num_channels, audio_codec_context->frame_size, audio_codec_context_get_audio_format(audio_codec_context)) != 0) {
fprintf(stderr, "Error: failed to get \"%s\" sound device\n", audio_input.name.c_str());
_exit(1);
}
+ audio_device.frame = create_audio_frame(audio_codec_context);
}
audio_devices.push_back(std::move(audio_device));
@@ -2032,7 +2036,6 @@ int main(int argc, char **argv) {
AudioTrack audio_track;
audio_track.codec_context = audio_codec_context;
- audio_track.frame = audio_frame;
audio_track.stream = audio_stream;
audio_track.audio_devices = std::move(audio_devices);
audio_track.graph = graph;
@@ -2138,14 +2141,14 @@ int main(int argc, char **argv) {
if(got_audio_data)
received_audio_time = this_audio_frame_time;
- int ret = av_frame_make_writable(audio_track.frame);
+ int ret = av_frame_make_writable(audio_device.frame);
if (ret < 0) {
fprintf(stderr, "Failed to make audio frame writable\n");
break;
}
// TODO: Is this |received_audio_time| really correct?
- int64_t num_missing_frames = std::round((this_audio_frame_time - received_audio_time) / target_audio_hz / (int64_t)audio_track.frame->nb_samples);
+ int64_t num_missing_frames = std::round((this_audio_frame_time - received_audio_time) / target_audio_hz / (int64_t)audio_device.frame->nb_samples);
if(got_audio_data)
num_missing_frames = std::max((int64_t)0, num_missing_frames - 1);
@@ -2164,29 +2167,29 @@ int main(int argc, char **argv) {
//audio_track.frame->data[0] = empty_audio;
received_audio_time = this_audio_frame_time;
if(needs_audio_conversion)
- swr_convert(swr, &audio_track.frame->data[0], audio_track.frame->nb_samples, (const uint8_t**)&empty_audio, audio_track.codec_context->frame_size);
+ swr_convert(swr, &audio_device.frame->data[0], audio_device.frame->nb_samples, (const uint8_t**)&empty_audio, audio_track.codec_context->frame_size);
else
- audio_track.frame->data[0] = empty_audio;
+ audio_device.frame->data[0] = empty_audio;
// TODO: Check if duplicate frame can be saved just by writing it with a different pts instead of sending it again
std::lock_guard<std::mutex> lock(audio_filter_mutex);
for(int i = 0; i < num_missing_frames; ++i) {
if(audio_track.graph) {
// TODO: av_buffersrc_add_frame
- if(av_buffersrc_write_frame(audio_device.src_filter_ctx, audio_track.frame) < 0) {
+ if(av_buffersrc_write_frame(audio_device.src_filter_ctx, audio_device.frame) < 0) {
fprintf(stderr, "Error: failed to add audio frame to filter\n");
}
} else {
- audio_track.frame->pts = (this_audio_frame_time - record_start_time) * (double)AV_TIME_BASE;
- const bool same_pts = audio_track.frame->pts == prev_pts;
- prev_pts = audio_track.frame->pts;
+ audio_device.frame->pts = (this_audio_frame_time - record_start_time) * (double)AV_TIME_BASE;
+ const bool same_pts = audio_device.frame->pts == prev_pts;
+ prev_pts = audio_device.frame->pts;
if(same_pts)
continue;
- ret = avcodec_send_frame(audio_track.codec_context, audio_track.frame);
+ ret = avcodec_send_frame(audio_track.codec_context, audio_device.frame);
if(ret >= 0) {
// TODO: Move to separate thread because this could write to network (for example when livestreaming)
- receive_frames(audio_track.codec_context, audio_track.stream_index, audio_track.stream, audio_track.frame->pts, av_format_context, record_start_time, frame_data_queue, replay_buffer_size_secs, frames_erased, write_output_mutex);
+ receive_frames(audio_track.codec_context, audio_track.stream_index, audio_track.stream, audio_device.frame->pts, av_format_context, record_start_time, frame_data_queue, replay_buffer_size_secs, frames_erased, write_output_mutex);
} else {
fprintf(stderr, "Failed to encode audio!\n");
}
@@ -2200,27 +2203,27 @@ int main(int argc, char **argv) {
if(got_audio_data) {
// TODO: Instead of converting audio, get float audio from alsa. Or does alsa do conversion internally to get this format?
if(needs_audio_conversion)
- swr_convert(swr, &audio_track.frame->data[0], audio_track.frame->nb_samples, (const uint8_t**)&sound_buffer, audio_track.codec_context->frame_size);
+ swr_convert(swr, &audio_device.frame->data[0], audio_device.frame->nb_samples, (const uint8_t**)&sound_buffer, audio_track.codec_context->frame_size);
else
- audio_track.frame->data[0] = (uint8_t*)sound_buffer;
+ audio_device.frame->data[0] = (uint8_t*)sound_buffer;
- audio_track.frame->pts = (this_audio_frame_time - record_start_time) * (double)AV_TIME_BASE;
- const bool same_pts = audio_track.frame->pts == prev_pts;
- prev_pts = audio_track.frame->pts;
+ audio_device.frame->pts = (this_audio_frame_time - record_start_time) * (double)AV_TIME_BASE;
+ const bool same_pts = audio_device.frame->pts == prev_pts;
+ prev_pts = audio_device.frame->pts;
if(same_pts)
continue;
if(audio_track.graph) {
std::lock_guard<std::mutex> lock(audio_filter_mutex);
// TODO: av_buffersrc_add_frame
- if(av_buffersrc_write_frame(audio_device.src_filter_ctx, audio_track.frame) < 0) {
+ if(av_buffersrc_write_frame(audio_device.src_filter_ctx, audio_device.frame) < 0) {
fprintf(stderr, "Error: failed to add audio frame to filter\n");
}
} else {
- ret = avcodec_send_frame(audio_track.codec_context, audio_track.frame);
+ ret = avcodec_send_frame(audio_track.codec_context, audio_device.frame);
if(ret >= 0) {
// TODO: Move to separate thread because this could write to network (for example when livestreaming)
- receive_frames(audio_track.codec_context, audio_track.stream_index, audio_track.stream, audio_track.frame->pts, av_format_context, record_start_time, frame_data_queue, replay_buffer_size_secs, frames_erased, write_output_mutex);
+ receive_frames(audio_track.codec_context, audio_track.stream_index, audio_track.stream, audio_device.frame->pts, av_format_context, record_start_time, frame_data_queue, replay_buffer_size_secs, frames_erased, write_output_mutex);
} else {
fprintf(stderr, "Failed to encode audio!\n");
}
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-06 23:37:48 +0000