diff options
Diffstat (limited to 'src/recorder/audio_codec.c')
| -rw-r--r-- | src/recorder/audio_codec.c | 219 |
1 files changed, 219 insertions, 0 deletions
diff --git a/src/recorder/audio_codec.c b/src/recorder/audio_codec.c new file mode 100644 index 0000000..e8f89e7 --- /dev/null +++ b/src/recorder/audio_codec.c @@ -0,0 +1,219 @@ +#include "../../include/recorder/audio_codec.h" +#include "../../include/ffmpeg_utils.h" +#include "../../include/log.h" + +#include <assert.h> +#include <math.h> + +#include <libavutil/opt.h> + +enum AVCodecID audio_codec_get_id(gsr_audio_codec audio_codec) { + switch(audio_codec) { + case GSR_AUDIO_CODEC_AAC: return AV_CODEC_ID_AAC; + case GSR_AUDIO_CODEC_OPUS: return AV_CODEC_ID_OPUS; + case GSR_AUDIO_CODEC_FLAC: return AV_CODEC_ID_FLAC; + } + assert(false); + return AV_CODEC_ID_AAC; +} + +enum AVSampleFormat audio_codec_get_sample_format(AVCodecContext *audio_codec_context, gsr_audio_codec audio_codec, const AVCodec *codec, bool mix_audio) { + (void)audio_codec_context; + switch(audio_codec) { + case GSR_AUDIO_CODEC_AAC: { + return AV_SAMPLE_FMT_FLTP; + } + case GSR_AUDIO_CODEC_OPUS: { + bool supports_s16 = false; + bool supports_flt = false; + + #if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(61, 15, 0) + for(size_t i = 0; codec->sample_fmts && codec->sample_fmts[i] != AV_SAMPLE_FMT_NONE; ++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; + } + } + #else + const enum AVSampleFormat *sample_fmts = NULL; + if(avcodec_get_supported_config(audio_codec_context, codec, AV_CODEC_CONFIG_SAMPLE_FORMAT, 0, (const void**)&sample_fmts, NULL) >= 0) { + if(sample_fmts) { + for(size_t i = 0; sample_fmts[i] != AV_SAMPLE_FMT_NONE; ++i) { + if(sample_fmts[i] == AV_SAMPLE_FMT_S16) { + supports_s16 = true; + } else if(sample_fmts[i] == AV_SAMPLE_FMT_FLT) { + supports_flt = true; + } + } + } else { + // What a dumb API. It returns NULL if all formats are supported + supports_s16 = true; + supports_flt = true; + } + } + #endif + + // Amix only works with float audio + if(mix_audio) + supports_s16 = false; + + if(!supports_s16 && !supports_flt) { + gsr_log(GSR_LOG_LEVEL_WARNING, "opus audio codec is chosen but your ffmpeg version does not support s16/flt sample format and performance might be slightly worse.\n" + " You can either rebuild ffmpeg with libopus instead of the built-in opus, use the flatpak version of gpu screen recorder or record with aac audio codec instead (-ac aac).\n" + " Falling back to fltp audio sample format instead."); + } + + if(supports_s16) + return AV_SAMPLE_FMT_S16; + else if(supports_flt) + return AV_SAMPLE_FMT_FLT; + else + return AV_SAMPLE_FMT_FLTP; + } + case GSR_AUDIO_CODEC_FLAC: { + return AV_SAMPLE_FMT_S32; + } + } + assert(false); + return AV_SAMPLE_FMT_FLTP; +} + +int64_t audio_codec_get_get_bitrate(gsr_audio_codec audio_codec) { + switch(audio_codec) { + case GSR_AUDIO_CODEC_AAC: return 160000; + case GSR_AUDIO_CODEC_OPUS: return 128000; + case GSR_AUDIO_CODEC_FLAC: return 128000; + } + assert(false); + return 128000; +} + +gsr_audio_format audio_codec_context_get_audio_format(const AVCodecContext *audio_codec_context) { + switch(audio_codec_context->sample_fmt) { + case AV_SAMPLE_FMT_FLT: return GSR_AUDIO_FORMAT_F32; + case AV_SAMPLE_FMT_FLTP: return GSR_AUDIO_FORMAT_S32; + case AV_SAMPLE_FMT_S16: return GSR_AUDIO_FORMAT_S16; + case AV_SAMPLE_FMT_S32: return GSR_AUDIO_FORMAT_S32; + default: return GSR_AUDIO_FORMAT_S16; + } +} + +enum AVSampleFormat audio_format_to_sample_format(const gsr_audio_format audio_format) { + switch(audio_format) { + case GSR_AUDIO_FORMAT_S16: return AV_SAMPLE_FMT_S16; + case GSR_AUDIO_FORMAT_S32: return AV_SAMPLE_FMT_S32; + case GSR_AUDIO_FORMAT_F32: return AV_SAMPLE_FMT_FLT; + } + assert(false); + return AV_SAMPLE_FMT_S16; +} + +AVCodecContext* create_audio_codec_context(int fps, gsr_audio_codec audio_codec, bool mix_audio, int64_t audio_bitrate) { + (void)fps; + const AVCodec *codec = avcodec_find_encoder(audio_codec_get_id(audio_codec)); + if (!codec) { + gsr_log(GSR_LOG_LEVEL_ERROR, "Could not find %s audio encoder", audio_codec_get_name(audio_codec)); + return NULL; + } + + 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(codec_context, audio_codec, codec, mix_audio); + codec_context->bit_rate = audio_bitrate == 0 ? audio_codec_get_get_bitrate(audio_codec) : audio_bitrate; + codec_context->sample_rate = GSR_AUDIO_SAMPLE_RATE; + if(audio_codec == GSR_AUDIO_CODEC_AAC) { +#if LIBAVCODEC_VERSION_MAJOR < 62 + codec_context->profile = FF_PROFILE_AAC_LOW; +#else + codec_context->profile = AV_PROFILE_AAC_LOW; +#endif + } +#if LIBAVCODEC_VERSION_MAJOR < 60 + codec_context->channel_layout = AV_CH_LAYOUT_STEREO; + codec_context->channels = 2; +#else + av_channel_layout_default(&codec_context->ch_layout, 2); +#endif + + codec_context->time_base.num = 1; + codec_context->time_base.den = codec_context->sample_rate; + codec_context->thread_count = 1; + codec_context->flags |= AV_CODEC_FLAG_GLOBAL_HEADER; + + return codec_context; +} + +bool open_audio(AVCodecContext *audio_codec_context, const char *ffmpeg_audio_opts) { + AVDictionary *options = NULL; + av_dict_set(&options, "strict", "experimental", 0); + + if(ffmpeg_audio_opts) + av_dict_parse_string(&options, ffmpeg_audio_opts, "=", ";", 0); + + int ret; + ret = avcodec_open2(audio_codec_context, audio_codec_context->codec, &options); + if(ret < 0) { + gsr_log(GSR_LOG_LEVEL_ERROR, "failed to open audio codec, reason: %s", gsr_av_error_to_string(ret)); + return false; + } + + return true; +} + +AVFrame* create_audio_frame(AVCodecContext *audio_codec_context) { + AVFrame *frame = av_frame_alloc(); + if(!frame) { + gsr_log(GSR_LOG_LEVEL_ERROR, "failed to allocate audio frame"); + return NULL; + } + + frame->sample_rate = audio_codec_context->sample_rate; + frame->nb_samples = audio_codec_context->frame_size; + frame->format = audio_codec_context->sample_fmt; +#if LIBAVCODEC_VERSION_MAJOR < 60 + frame->channels = audio_codec_context->channels; + frame->channel_layout = audio_codec_context->channel_layout; +#else + av_channel_layout_copy(&frame->ch_layout, &audio_codec_context->ch_layout); +#endif + + int ret = av_frame_get_buffer(frame, 0); + if(ret < 0) { + gsr_log(GSR_LOG_LEVEL_ERROR, "failed to allocate audio data buffers, reason: %s", gsr_av_error_to_string(ret)); + av_frame_free(&frame); + return NULL; + } + + return frame; +} + +double audio_codec_get_desired_delay(gsr_audio_codec audio_codec, int fps) { + const double fps_inv = 1.0 / (double)fps; + const double base = 0.01 + 1.0/165.0; + switch(audio_codec) { + case GSR_AUDIO_CODEC_OPUS: + return fmax(0.0, base - fps_inv); + case GSR_AUDIO_CODEC_AAC: + return fmax(0.0, (base + 0.008) * 2.0 - fps_inv); + case GSR_AUDIO_CODEC_FLAC: + // TODO: Test + return fmax(0.0, base - fps_inv); + } + assert(false); + return fmax(0.0, base - fps_inv); +} + +int audio_codec_get_frame_size(gsr_audio_codec audio_codec) { + switch(audio_codec) { + case GSR_AUDIO_CODEC_AAC: return 1024; + case GSR_AUDIO_CODEC_OPUS: return 960; + case GSR_AUDIO_CODEC_FLAC: + assert(false); + return 1024; + } + assert(false); + return 1024; +} |