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authordec05eba <dec05eba@protonmail.com>2023-03-04 16:02:48 +0100
committerdec05eba <dec05eba@protonmail.com>2023-03-04 16:02:48 +0100
commit8ca1ac70e8466d7aba0a57236a1d068939540a3f (patch)
tree7d1596783e377f7975ed8c8af570db0decb77ad4 /src
parentc956cd4be3a4e81d1b50665debf28395a87f393f (diff)
Support opus flt and fltp
Diffstat (limited to 'src')
-rw-r--r--src/main.cpp64
-rw-r--r--src/sound.cpp2
2 files changed, 44 insertions, 22 deletions
diff --git a/src/main.cpp b/src/main.cpp
index 2444aad..d007b0d 100644
--- a/src/main.cpp
+++ b/src/main.cpp
@@ -235,11 +235,37 @@ static AVCodecID audio_codec_get_id(AudioCodec audio_codec) {
return AV_CODEC_ID_AAC;
}
-static AVSampleFormat audio_codec_get_sample_format(AudioCodec audio_codec) {
+static AVSampleFormat audio_codec_get_sample_format(AudioCodec audio_codec, const AVCodec *codec) {
switch(audio_codec) {
- case AudioCodec::AAC: return AV_SAMPLE_FMT_FLTP;
- case AudioCodec::OPUS: return AV_SAMPLE_FMT_S16;
- case AudioCodec::FLAC: return AV_SAMPLE_FMT_S32;
+ case AudioCodec::AAC: {
+ return AV_SAMPLE_FMT_FLTP;
+ }
+ case AudioCodec::OPUS: {
+ bool supports_s16 = false;
+ bool supports_flt = false;
+
+ for(size_t i = 0; codec->sample_fmts && codec->sample_fmts[i] != -1; ++i) {
+ if(codec->sample_fmts[i] == AV_SAMPLE_FMT_S16) {
+ supports_s16 = true;
+ } else if(codec->sample_fmts[i] == AV_SAMPLE_FMT_FLT) {
+ supports_flt = true;
+ }
+ }
+
+ if(!supports_s16 && !supports_flt) {
+ fprintf(stderr, "Warning: opus audio codec is chosen but your ffmpeg version does not support s16/flt sample format and performance might be slightly worse. You can either rebuild ffmpeg with libopus instead of the built-in opus, use the flatpak version of gpu screen recorder or record with flac audio codec instead (-ac flac). Falling back to fltp audio sample format instead.\n");
+ }
+
+ if(supports_s16)
+ return AV_SAMPLE_FMT_S16;
+ else if(supports_flt)
+ return AV_SAMPLE_FMT_FLT;
+ else
+ return AV_SAMPLE_FMT_FLTP;
+ }
+ case AudioCodec::FLAC: {
+ return AV_SAMPLE_FMT_S32;
+ }
}
assert(false);
return AV_SAMPLE_FMT_FLTP;
@@ -255,20 +281,21 @@ static int64_t audio_codec_get_get_bitrate(AudioCodec audio_codec) {
return 96000;
}
-static AudioFormat audio_codec_get_audio_format(AudioCodec audio_codec) {
- switch(audio_codec) {
- case AudioCodec::AAC: return S32;
- case AudioCodec::OPUS: return S16;
- case AudioCodec::FLAC: return S32;
+static AudioFormat audio_codec_context_get_audio_format(const AVCodecContext *audio_codec_context) {
+ switch(audio_codec_context->sample_fmt) {
+ case AV_SAMPLE_FMT_FLT: return F32;
+ case AV_SAMPLE_FMT_FLTP: return S32;
+ case AV_SAMPLE_FMT_S16: return S16;
+ case AV_SAMPLE_FMT_S32: return S32;
+ default: return S16;
}
- assert(false);
- return S32;
}
static AVSampleFormat audio_format_to_sample_format(const AudioFormat audio_format) {
switch(audio_format) {
case S16: return AV_SAMPLE_FMT_S16;
case S32: return AV_SAMPLE_FMT_S32;
+ case F32: return AV_SAMPLE_FMT_FLT;
}
assert(false);
return AV_SAMPLE_FMT_S16;
@@ -281,16 +308,11 @@ static AVCodecContext* create_audio_codec_context(int fps, AudioCodec audio_code
exit(1);
}
- fprintf(stderr, "Audio codec: %s, supported sample formats:\n", audio_codec_get_name(audio_codec));
- for(size_t i = 0; codec->sample_fmts && codec->sample_fmts[i] != -1; ++i) {
- fprintf(stderr, " %zu: %s\n", i, av_get_sample_fmt_name(codec->sample_fmts[i]));
- }
-
AVCodecContext *codec_context = avcodec_alloc_context3(codec);
assert(codec->type == AVMEDIA_TYPE_AUDIO);
codec_context->codec_id = codec->id;
- codec_context->sample_fmt = audio_codec_get_sample_format(audio_codec);
+ codec_context->sample_fmt = audio_codec_get_sample_format(audio_codec, codec);
codec_context->bit_rate = audio_codec_get_get_bitrate(audio_codec);
codec_context->sample_rate = 48000;
if(audio_codec == AudioCodec::AAC)
@@ -1055,8 +1077,6 @@ int main(int argc, char **argv) {
usage();
}
- const AudioFormat audio_format = audio_codec_get_audio_format(audio_codec);
-
const Arg &audio_input_arg = args["-a"];
const std::vector<AudioInput> audio_inputs = get_pulseaudio_inputs();
std::vector<MergedAudioInputs> requested_audio_inputs;
@@ -1482,7 +1502,7 @@ int main(int argc, char **argv) {
audio_device.sound_device.handle = NULL;
audio_device.sound_device.frames = 0;
} 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_format) != 0) {
+ 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);
}
@@ -1560,8 +1580,8 @@ int main(int argc, char **argv) {
for(AudioTrack &audio_track : audio_tracks) {
for(AudioDevice &audio_device : audio_track.audio_devices) {
- audio_device.thread = std::thread([record_start_time, replay_buffer_size_secs, &frame_data_queue, &frames_erased, &audio_track, empty_audio, &audio_device, &audio_filter_mutex, &write_output_mutex, audio_format](AVFormatContext *av_format_context) mutable {
- const AVSampleFormat sound_device_sample_format = audio_format_to_sample_format(audio_format);
+ audio_device.thread = std::thread([record_start_time, replay_buffer_size_secs, &frame_data_queue, &frames_erased, &audio_track, empty_audio, &audio_device, &audio_filter_mutex, &write_output_mutex](AVFormatContext *av_format_context) mutable {
+ const AVSampleFormat sound_device_sample_format = audio_format_to_sample_format(audio_codec_context_get_audio_format(audio_track.codec_context));
const bool needs_audio_conversion = audio_track.codec_context->sample_fmt != sound_device_sample_format;
SwrContext *swr = nullptr;
if(needs_audio_conversion) {
diff --git a/src/sound.cpp b/src/sound.cpp
index 24e979b..762d962 100644
--- a/src/sound.cpp
+++ b/src/sound.cpp
@@ -231,6 +231,7 @@ static pa_sample_format_t audio_format_to_pulse_audio_format(AudioFormat audio_f
switch(audio_format) {
case S16: return PA_SAMPLE_S16LE;
case S32: return PA_SAMPLE_S32LE;
+ case F32: return PA_SAMPLE_FLOAT32LE;
}
assert(false);
return PA_SAMPLE_S16LE;
@@ -240,6 +241,7 @@ static int audio_format_to_get_bytes_per_sample(AudioFormat audio_format) {
switch(audio_format) {
case S16: return 2;
case S32: return 4;
+ case F32: return 4;
}
assert(false);
return 2;