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Planarize audio.

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This commit is contained in:
Themaister 2012-12-09 00:20:04 +01:00
parent ff1557d6e8
commit 47fca83bb1
1 changed files with 61 additions and 8 deletions
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65
record/ffemu.c
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@ -101,6 +101,7 @@ struct ff_audio_info
// Use either S16 or (planar) float for simplicity.
rarch_resampler_t *resampler;
bool use_float;
bool is_planar;
unsigned sample_size;
float *float_conv;
@ -112,6 +113,9 @@ struct ff_audio_info
int16_t *fixed_conv;
size_t fixed_conv_frames;
void *planar_buf;
size_t planar_buf_frames;
double ratio;
};
@ -180,27 +184,30 @@ static void ffemu_audio_resolve_format(struct ff_audio_info *audio, const AVCode
{
audio->codec->sample_fmt = AV_SAMPLE_FMT_FLTP;
audio->use_float = true;
audio->is_planar = true;
RARCH_LOG("[FFmpeg]: Using sample format FLTP.\n");
}
else if (ffemu_codec_has_sample_format(AV_SAMPLE_FMT_FLT, codec->sample_fmts))
{
audio->codec->sample_fmt = AV_SAMPLE_FMT_FLT;
audio->use_float = true;
audio->is_planar = false;
RARCH_LOG("[FFmpeg]: Using sample format FLT.\n");
}
else if (ffemu_codec_has_sample_format(AV_SAMPLE_FMT_S16P, codec->sample_fmts))
{
audio->codec->sample_fmt = AV_SAMPLE_FMT_S16P;
audio->use_float = false;
audio->is_planar = true;
RARCH_LOG("[FFmpeg]: Using sample format S16P.\n");
}
else if (ffemu_codec_has_sample_format(AV_SAMPLE_FMT_S16, codec->sample_fmts))
{
audio->codec->sample_fmt = AV_SAMPLE_FMT_S16;
audio->use_float = false;
audio->is_planar = false;
RARCH_LOG("[FFmpeg]: Using sample format S16.\n");
}
audio->sample_size = audio->use_float ? sizeof(float) : sizeof(int16_t);
}
@ -682,6 +689,7 @@ void ffemu_free(ffemu_t *handle)
av_free(handle->audio.float_conv);
av_free(handle->audio.resample_out);
av_free(handle->audio.fixed_conv);
av_free(handle->audio.planar_buf);
free(handle);
}
@ -866,6 +874,47 @@ static bool ffemu_push_video_thread(ffemu_t *handle, const struct ffemu_video_da
return true;
}
static void planarize_float(float *out, const float *in, size_t frames)
{
for (size_t i = 0; i < frames; i++)
{
out[i] = in[2 * i + 0];
out[i + frames] = in[2 * i + 1];
}
}
static void planarize_s16(int16_t *out, const int16_t *in, size_t frames)
{
for (size_t i = 0; i < frames; i++)
{
out[i] = in[2 * i + 0];
out[i + frames] = in[2 * i + 1];
}
}
static void planarize_audio(ffemu_t *handle)
{
if (!handle->audio.is_planar)
return;
if (handle->audio.frames_in_buffer > handle->audio.planar_buf_frames)
{
handle->audio.planar_buf = av_realloc(handle->audio.planar_buf,
handle->audio.resample_out_frames * handle->params.channels * handle->audio.sample_size);
if (!handle->audio.planar_buf)
return;
handle->audio.planar_buf_frames = handle->audio.frames_in_buffer;
}
if (handle->audio.use_float)
planarize_float((float*)handle->audio.planar_buf,
(const float*)handle->audio.buffer, handle->audio.frames_in_buffer);
else
planarize_s16((int16_t*)handle->audio.planar_buf,
(const int16_t*)handle->audio.buffer, handle->audio.frames_in_buffer);
}
static bool encode_audio(ffemu_t *handle, AVPacket *pkt, bool dry)
{
av_init_packet(pkt);
@ -877,16 +926,20 @@ static bool encode_audio(ffemu_t *handle, AVPacket *pkt, bool dry)
return false;
frame->nb_samples = handle->audio.frames_in_buffer;
frame->format = handle->audio.codec->sample_fmt;
frame->channel_layout = handle->audio.codec->channel_layout;
frame->pts = handle->audio.frame_cnt;
int samples_size = frame->nb_samples *
handle->audio.codec->channels *
handle->audio.sample_size;
planarize_audio(handle);
int samples_size = av_samples_get_buffer_size(NULL, handle->audio.codec->channels,
handle->audio.frames_in_buffer,
handle->audio.codec->sample_fmt, 0);
avcodec_fill_audio_frame(frame, handle->audio.codec->channels,
handle->audio.codec->sample_fmt,
handle->audio.buffer,
samples_size, 1);
handle->audio.is_planar ? (uint8_t*)handle->audio.planar_buf : handle->audio.buffer,
samples_size, 0);
int got_packet = 0;
if (avcodec_encode_audio2(handle->audio.codec,