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Remove specialized code for resampling to allow for broader variety of sample formats

This commit is contained in:
Jeffrey Pfau 2014-10-27 22:57:26 -07:00
parent aed170b670
commit 4fb82e4b80
1 changed files with 49 additions and 63 deletions
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78
src/platform/ffmpeg/ffmpeg-encoder.c
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@ -31,6 +31,19 @@ void FFmpegEncoderInit(struct FFmpegEncoder* encoder) {
}
bool FFmpegEncoderSetAudio(struct FFmpegEncoder* encoder, const char* acodec, unsigned abr) {
static const struct {
int format;
int priority;
} priorities[] = {
{ AV_SAMPLE_FMT_S16, 0 },
{ AV_SAMPLE_FMT_S16P, 1 },
{ AV_SAMPLE_FMT_S32, 2 },
{ AV_SAMPLE_FMT_S32P, 2 },
{ AV_SAMPLE_FMT_FLT, 3 },
{ AV_SAMPLE_FMT_FLTP, 3 },
{ AV_SAMPLE_FMT_DBL, 4 },
{ AV_SAMPLE_FMT_DBLP, 4 }
};
AVCodec* codec = avcodec_find_encoder_by_name(acodec);
if (!codec) {
return false;
@ -40,11 +53,15 @@ bool FFmpegEncoderSetAudio(struct FFmpegEncoder* encoder, const char* acodec, un
return false;
}
size_t i;
size_t j;
int priority = INT_MAX;
encoder->sampleFormat = AV_SAMPLE_FMT_NONE;
for (i = 0; codec->sample_fmts[i] != AV_SAMPLE_FMT_NONE; ++i) {
if (codec->sample_fmts[i] == AV_SAMPLE_FMT_S16 || codec->sample_fmts[i] == AV_SAMPLE_FMT_S16P) {
for (j = 0; j < sizeof(priorities) / sizeof(*priorities); ++j) {
if (codec->sample_fmts[i] == priorities[j].format && priority > priorities[j].priority) {
priority = priorities[j].priority;
encoder->sampleFormat = codec->sample_fmts[i];
break;
}
}
}
if (encoder->sampleFormat == AV_SAMPLE_FMT_NONE) {
@ -70,7 +87,7 @@ bool FFmpegEncoderSetAudio(struct FFmpegEncoder* encoder, const char* acodec, un
}
bool FFmpegEncoderSetVideo(struct FFmpegEncoder* encoder, const char* vcodec, unsigned vbr) {
static struct {
static const struct {
enum AVPixelFormat format;
int priority;
} priorities[] = {
@ -148,27 +165,24 @@ bool FFmpegEncoderOpen(struct FFmpegEncoder* encoder, const char* outfile) {
encoder->audio->channel_layout = AV_CH_LAYOUT_STEREO;
encoder->audio->sample_rate = encoder->sampleRate;
encoder->audio->sample_fmt = encoder->sampleFormat;
avcodec_open2(encoder->audio, acodec, 0);
AVDictionary* opts = 0;
av_dict_set_int(&opts, "strict", FF_COMPLIANCE_EXPERIMENTAL, 0);
avcodec_open2(encoder->audio, acodec, &opts);
av_dict_free(&opts);
encoder->audioFrame = av_frame_alloc();
encoder->audioFrame->nb_samples = encoder->audio->frame_size;
encoder->audioFrame->format = encoder->audio->sample_fmt;
encoder->audioFrame->pts = 0;
if (encoder->sampleRate != PREFERRED_SAMPLE_RATE) {
encoder->resampleContext = avresample_alloc_context();
av_opt_set_int(encoder->resampleContext, "in_channel_layout", AV_CH_LAYOUT_STEREO, 0);
av_opt_set_int(encoder->resampleContext, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
av_opt_set_int(encoder->resampleContext, "in_sample_rate", PREFERRED_SAMPLE_RATE, 0);
av_opt_set_int(encoder->resampleContext, "out_sample_rate", encoder->sampleRate, 0);
av_opt_set_int(encoder->resampleContext, "in_sample_fmt", encoder->sampleFormat, 0);
av_opt_set_int(encoder->resampleContext, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
av_opt_set_int(encoder->resampleContext, "out_sample_fmt", encoder->sampleFormat, 0);
avresample_open(encoder->resampleContext);
encoder->audioBufferSize = (encoder->audioFrame->nb_samples * PREFERRED_SAMPLE_RATE / encoder->sampleRate) * 4;
encoder->audioBuffer = av_malloc(encoder->audioBufferSize);
} else {
encoder->resampleContext = 0;
encoder->audioBufferSize = 0;
encoder->audioBuffer = 0;
}
encoder->postaudioBufferSize = av_samples_get_buffer_size(0, encoder->audio->channels, encoder->audio->frame_size, encoder->audio->sample_fmt, 0);
encoder->postaudioBuffer = av_malloc(encoder->postaudioBufferSize);
avcodec_fill_audio_frame(encoder->audioFrame, encoder->audio->channels, encoder->audio->sample_fmt, (const uint8_t*) encoder->postaudioBuffer, encoder->postaudioBufferSize, 0);
@ -237,53 +251,25 @@ void _ffmpegPostAudioFrame(struct GBAAVStream* stream, int32_t left, int32_t rig
}
av_frame_make_writable(encoder->audioFrame);
uint16_t* buffers[2];
int stride;
bool planar = av_sample_fmt_is_planar(encoder->audio->sample_fmt);
if (encoder->resampleContext) {
buffers[0] = (uint16_t*) encoder->audioBuffer;
if (planar) {
stride = 1;
buffers[1] = &buffers[0][encoder->audioBufferSize / 4];
} else {
stride = 2;
buffers[1] = &buffers[0][1];
}
} else {
buffers[0] = (uint16_t*) encoder->postaudioBuffer;
if (planar) {
stride = 1;
buffers[1] = &buffers[0][encoder->postaudioBufferSize / 4];
} else {
stride = 2;
buffers[1] = &buffers[0][1];
}
}
buffers[0][encoder->currentAudioSample * stride] = left;
buffers[1][encoder->currentAudioSample * stride] = right;
encoder->audioBuffer[encoder->currentAudioSample * 2] = left;
encoder->audioBuffer[encoder->currentAudioSample * 2 + 1] = right;
++encoder->currentAudioFrame;
++encoder->currentAudioSample;
if (encoder->resampleContext) {
if ((encoder->currentAudioSample * 4) < encoder->audioBufferSize) {
return;
}
encoder->currentAudioSample = 0;
int channelSize = 2 * av_get_bytes_per_sample(encoder->audio->sample_fmt);
avresample_convert(encoder->resampleContext,
0, 0, encoder->postaudioBufferSize / 4,
(uint8_t**) buffers, 0, encoder->audioBufferSize / 4);
if ((ssize_t) avresample_available(encoder->resampleContext) < (ssize_t) encoder->postaudioBufferSize / 4) {
0, 0, encoder->postaudioBufferSize / channelSize,
(uint8_t**) &encoder->audioBuffer, 0, encoder->audioBufferSize / 4);
if ((ssize_t) avresample_available(encoder->resampleContext) < (ssize_t) encoder->postaudioBufferSize / channelSize) {
return;
}
avresample_read(encoder->resampleContext, encoder->audioFrame->data, encoder->postaudioBufferSize / 4);
} else {
if ((encoder->currentAudioSample * 4) < encoder->postaudioBufferSize) {
return;
}
encoder->currentAudioSample = 0;
}
avresample_read(encoder->resampleContext, encoder->audioFrame->data, encoder->postaudioBufferSize / channelSize);
AVRational timeBase = { 1, PREFERRED_SAMPLE_RATE };
encoder->audioFrame->pts = av_rescale_q(encoder->currentAudioFrame, timeBase, encoder->audioStream->time_base);