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https://github.com/libretro/RetroArch
synced 2025-02-01 20:54:10 +00:00
Handle floating point/fixed formats seamlessly.
This commit is contained in:
Themaister
parent
dcf505f0ea
commit
64d3d2185c
102
record/ffemu.c
102
record/ffemu.c
@ -103,10 +103,17 @@ struct ff_audio_info
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bool use_float;
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unsigned sample_size;
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enum AVSampleFormat fill_format;
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float *float_conv;
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size_t float_conv_frames;
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float *resample_out;
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size_t resample_out_frames;
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int16_t *fixed_conv;
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size_t fixed_conv_frames;
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double ratio;
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};
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@ -159,6 +166,51 @@ struct ffemu
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volatile bool can_sleep;
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};
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static bool ffemu_codec_has_sample_format(enum AVSampleFormat fmt, const enum AVSampleFormat *fmts)
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{
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for (unsigned i = 0; fmts[i] != AV_SAMPLE_FMT_NONE; i++)
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if (fmt == fmts[i])
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return true;
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return false;
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}
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static void ffemu_audio_resolve_format(struct ff_audio_info *audio, AVCodec *codec)
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{
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audio->codec->sample_fmt = AV_SAMPLE_FMT_NONE;
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audio->fill_format = AV_SAMPLE_FMT_NONE;
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if (ffemu_codec_has_sample_format(AV_SAMPLE_FMT_FLTP, codec->sample_fmts))
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{
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audio->codec->sample_fmt = AV_SAMPLE_FMT_FLT;
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audio->fill_format = AV_SAMPLE_FMT_FLTP;
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audio->use_float = true;
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RARCH_LOG("[FFmpeg]: Using sample format FLTP.\n");
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}
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else if (ffemu_codec_has_sample_format(AV_SAMPLE_FMT_FLT, codec->sample_fmts))
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{
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audio->codec->sample_fmt = AV_SAMPLE_FMT_FLT;
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audio->fill_format = AV_SAMPLE_FMT_FLT;
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audio->use_float = true;
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RARCH_LOG("[FFmpeg]: Using sample format FLT.\n");
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}
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else if (ffemu_codec_has_sample_format(AV_SAMPLE_FMT_S16P, codec->sample_fmts))
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{
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audio->codec->sample_fmt = AV_SAMPLE_FMT_S16;
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audio->fill_format = AV_SAMPLE_FMT_S16P;
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audio->use_float = false;
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RARCH_LOG("[FFmpeg]: Using sample format S16P.\n");
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}
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else if (ffemu_codec_has_sample_format(AV_SAMPLE_FMT_S16, codec->sample_fmts))
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{
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audio->codec->sample_fmt = AV_SAMPLE_FMT_S16;
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audio->fill_format = AV_SAMPLE_FMT_S16;
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audio->use_float = false;
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RARCH_LOG("[FFmpeg]: Using sample format S16.\n");
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}
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audio->sample_size = audio->use_float ? sizeof(float) : sizeof(int16_t);
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}
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static bool ffemu_init_audio(ffemu_t *handle)
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{
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struct ff_config_param *params = &handle->config;
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@ -176,23 +228,22 @@ static bool ffemu_init_audio(ffemu_t *handle)
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audio->codec = avcodec_alloc_context3(codec);
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audio->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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audio->codec->channels = param->channels;
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audio->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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audio->codec->channels = param->channels;
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audio->codec->channel_layout = AV_CH_LAYOUT_STEREO;
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ffemu_audio_resolve_format(audio, codec);
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if (params->sample_rate)
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{
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RARCH_LOG("[FFmpeg]: Using FLT sampling format.\n");
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audio->codec->sample_fmt = AV_SAMPLE_FMT_FLT;
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audio->ratio = (double)params->sample_rate / param->samplerate;
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audio->codec->sample_rate = params->sample_rate;
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audio->codec->time_base = av_d2q(1.0 / params->sample_rate, 1000000);
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audio->resampler = resampler_new();
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audio->use_float = true;
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}
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else
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{
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RARCH_LOG("[FFmpeg]: Using S16 sampling format.\n");
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audio->codec->sample_fmt = AV_SAMPLE_FMT_S16;
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audio->codec->sample_rate = (int)roundf(param->samplerate);
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audio->codec->time_base = av_d2q(1.0 / param->samplerate, 1000000);
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@ -206,8 +257,6 @@ static bool ffemu_init_audio(ffemu_t *handle)
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else if (params->audio_bit_rate)
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audio->codec->bit_rate = params->audio_bit_rate;
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audio->sample_size = audio->use_float ? sizeof(float) : sizeof(int16_t);
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// Allow experimental codecs.
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audio->codec->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
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@ -574,8 +623,7 @@ void ffemu_free(ffemu_t *handle)
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av_free(handle->audio.codec);
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}
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if (handle->audio.buffer)
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av_free(handle->audio.buffer);
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av_free(handle->audio.buffer);
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if (handle->video.codec)
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{
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@ -583,11 +631,8 @@ void ffemu_free(ffemu_t *handle)
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av_free(handle->video.codec);
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}
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if (handle->video.conv_frame)
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av_free(handle->video.conv_frame);
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if (handle->video.conv_frame_buf)
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av_free(handle->video.conv_frame_buf);
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av_free(handle->video.conv_frame);
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av_free(handle->video.conv_frame_buf);
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scaler_ctx_gen_reset(&handle->video.scaler);
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@ -603,8 +648,10 @@ void ffemu_free(ffemu_t *handle)
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if (handle->audio.resampler)
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resampler_free(handle->audio.resampler);
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av_free(handle->audio.float_conv);
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av_free(handle->audio.resample_out);
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av_free(handle->audio.fixed_conv);
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free(handle);
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}
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@ -807,7 +854,7 @@ static bool encode_audio(ffemu_t *handle, AVPacket *pkt, bool dry)
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handle->audio.sample_size;
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avcodec_fill_audio_frame(frame, handle->audio.codec->channels,
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handle->audio.use_float ? AV_SAMPLE_FMT_FLTP : AV_SAMPLE_FMT_S16,
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handle->audio.fill_format,
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handle->audio.buffer,
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samples_size, 1);
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@ -850,7 +897,7 @@ static bool encode_audio(ffemu_t *handle, AVPacket *pkt, bool dry)
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static void ffemu_audio_resample(ffemu_t *handle, struct ffemu_audio_data *data)
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{
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if (!handle->audio.use_float)
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if (!handle->audio.use_float && !handle->audio.resampler)
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return;
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if (data->frames > handle->audio.float_conv_frames)
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@ -869,11 +916,21 @@ static void ffemu_audio_resample(ffemu_t *handle, struct ffemu_audio_data *data)
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handle->audio.resample_out_frames * handle->params.channels * sizeof(float));
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if (!handle->audio.resample_out)
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return;
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handle->audio.fixed_conv_frames = max(handle->audio.resample_out_frames, handle->audio.float_conv_frames);
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handle->audio.fixed_conv = (int16_t*)av_realloc(handle->audio.fixed_conv,
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handle->audio.fixed_conv_frames * handle->params.channels * sizeof(int16_t));
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if (!handle->audio.fixed_conv)
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return;
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}
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audio_convert_s16_to_float(handle->audio.float_conv, (const int16_t*)data->data, data->frames * handle->params.channels, 1.0);
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if (handle->audio.use_float || handle->audio.resampler)
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{
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audio_convert_s16_to_float(handle->audio.float_conv,
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(const int16_t*)data->data, data->frames * handle->params.channels, 1.0);
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data->data = handle->audio.float_conv;
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}
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data->data = handle->audio.float_conv;
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if (handle->audio.resampler)
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{
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// It's always two channels ...
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@ -886,6 +943,13 @@ static void ffemu_audio_resample(ffemu_t *handle, struct ffemu_audio_data *data)
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resampler_process(handle->audio.resampler, &info);
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data->data = handle->audio.resample_out;
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data->frames = info.output_frames;
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if (!handle->audio.use_float)
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{
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audio_convert_float_to_s16(handle->audio.fixed_conv, handle->audio.resample_out,
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data->frames * handle->params.channels);
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data->data = handle->audio.fixed_conv;
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}
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}
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}
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