== Sequencer ==
This fixes: [#18507] Wrong audio mixdown Also: you can change output sample rate while blender is running and the audio device get's reopened automatically. Subframe-precision seeking was also broken for some releases... (shame on me)
This commit is contained in:
Peter Schlaile
parent
af42ee43be
commit
2a3627e338
@ -216,6 +216,97 @@ long sound_hdaudio_get_duration(struct hdaudio * hdaudio, double frame_rate)
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}
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#ifdef WITH_FFMPEG
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#define RESAMPLE_FILL_RATIO (7.0/4.0)
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static void sound_hdaudio_run_resampler_seek(
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struct hdaudio * hdaudio)
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{
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int in_frame_size = (long long) hdaudio->sample_rate
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* hdaudio->frame_duration / AV_TIME_BASE;
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hdaudio->resample_samples_in = in_frame_size * RESAMPLE_FILL_RATIO;
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hdaudio->resample_samples_written
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= audio_resample(hdaudio->resampler,
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hdaudio->resample_cache,
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hdaudio->decode_cache_zero,
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in_frame_size * RESAMPLE_FILL_RATIO);
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}
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static void sound_hdaudio_run_resampler_continue(
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struct hdaudio * hdaudio)
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{
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int target_rate = hdaudio->target_rate;
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int target_channels = hdaudio->target_channels;
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int frame_size = (long long) target_rate
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* hdaudio->frame_duration / AV_TIME_BASE;
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int in_frame_size = (long long) hdaudio->sample_rate
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* hdaudio->frame_duration / AV_TIME_BASE;
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int reuse_tgt = (hdaudio->resample_samples_written
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- frame_size) * target_channels;
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int reuse_src = (hdaudio->resample_samples_in
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- in_frame_size) * hdaudio->channels;
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int next_samples_in =
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in_frame_size * RESAMPLE_FILL_RATIO
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- reuse_src / hdaudio->channels;
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memmove(hdaudio->resample_cache,
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hdaudio->resample_cache + frame_size * target_channels,
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reuse_tgt * sizeof(short));
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hdaudio->resample_samples_written
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= audio_resample(
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hdaudio->resampler,
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hdaudio->resample_cache + reuse_tgt,
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hdaudio->decode_cache_zero + reuse_src,
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next_samples_in)
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+ reuse_tgt / target_channels;
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hdaudio->resample_samples_in = next_samples_in
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+ reuse_src / hdaudio->channels;
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}
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static void sound_hdaudio_init_resampler(
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struct hdaudio * hdaudio,
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int frame_position, int target_rate, int target_channels)
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{
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int frame_size = (long long) target_rate
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* hdaudio->frame_duration / AV_TIME_BASE;
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if (hdaudio->resampler &&
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(hdaudio->target_rate != target_rate
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|| hdaudio->target_channels != target_channels)) {
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audio_resample_close(hdaudio->resampler);
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hdaudio->resampler = 0;
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}
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if (!hdaudio->resampler) {
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hdaudio->resampler = av_audio_resample_init(
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target_channels, hdaudio->channels,
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target_rate, hdaudio->sample_rate,
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SAMPLE_FMT_S16, SAMPLE_FMT_S16,
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16, 10, 0, 0.8);
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hdaudio->target_rate = target_rate;
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hdaudio->target_channels = target_channels;
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if (hdaudio->resample_cache) {
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MEM_freeN(hdaudio->resample_cache);
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}
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hdaudio->resample_cache = (short*) MEM_mallocN(
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(long long)
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hdaudio->target_channels
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* frame_size * 2
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* sizeof(short),
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"hdaudio resample cache");
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if (frame_position == hdaudio->frame_position ||
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frame_position == hdaudio->frame_position + 1) {
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sound_hdaudio_run_resampler_seek(hdaudio);
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}
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}
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}
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static void sound_hdaudio_extract_small_block(
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struct hdaudio * hdaudio,
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short * target_buffer,
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@ -240,40 +331,9 @@ static void sound_hdaudio_extract_small_block(
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if (hdaudio == 0) return;
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if (rate_conversion) {
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if (hdaudio->resampler &&
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(hdaudio->target_rate != target_rate
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|| hdaudio->target_channels != target_channels)) {
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audio_resample_close(hdaudio->resampler);
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hdaudio->resampler = 0;
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}
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if (!hdaudio->resampler) {
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hdaudio->resampler = audio_resample_init(
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target_channels, hdaudio->channels,
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target_rate, hdaudio->sample_rate);
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hdaudio->target_rate = target_rate;
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hdaudio->target_channels = target_channels;
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if (hdaudio->resample_cache) {
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MEM_freeN(hdaudio->resample_cache);
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}
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hdaudio->resample_cache = (short*) MEM_mallocN(
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(long long)
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hdaudio->target_channels
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* frame_size * 2
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* sizeof(short),
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"hdaudio resample cache");
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if (frame_position == hdaudio->frame_position) {
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hdaudio->resample_samples_in =
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in_frame_size * 7 / 4;
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hdaudio->resample_samples_written
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= audio_resample(
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hdaudio->resampler,
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hdaudio->resample_cache,
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hdaudio->decode_cache_zero,
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in_frame_size * 7 / 4);
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}
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}
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sound_hdaudio_init_resampler(
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hdaudio, frame_position,
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target_rate, target_channels);
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}
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if (frame_position == hdaudio->frame_position + 1
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@ -347,36 +407,11 @@ static void sound_hdaudio_extract_small_block(
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}
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}
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if (rate_conversion) {
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int written = hdaudio->resample_samples_written
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* target_channels;
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int ofs = target_channels * frame_size;
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int recycle = written - ofs;
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int next_in = in_frame_size
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+ (3.0/4.0
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- (double) recycle / (double)
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(frame_size * target_channels)
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) * in_frame_size;
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memmove(hdaudio->resample_cache,
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hdaudio->resample_cache + ofs,
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recycle * sizeof(short));
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hdaudio->resample_samples_written
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= audio_resample(
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hdaudio->resampler,
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hdaudio->resample_cache + recycle,
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hdaudio->decode_cache_zero
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+ hdaudio->resample_samples_in
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* hdaudio->channels
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- bl_size,
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next_in)
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+ recycle / target_channels;
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hdaudio->resample_samples_in = next_in;
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}
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hdaudio->decode_pos = decode_pos;
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if (rate_conversion) {
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sound_hdaudio_run_resampler_continue(hdaudio);
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}
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}
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if (frame_position != hdaudio->frame_position) {
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@ -390,6 +425,7 @@ static void sound_hdaudio_extract_small_block(
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* hdaudio->frame_duration / AV_TIME_BASE;
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long long seek_pos;
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int decode_cache_zero_init = 0;
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hdaudio->frame_position = frame_position;
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@ -435,8 +471,7 @@ static void sound_hdaudio_extract_small_block(
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audio_pkt_data = packet.data;
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audio_pkt_size = packet.size;
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if (!hdaudio->decode_cache_zero
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&& audio_pkt_size > 0) {
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if (!decode_cache_zero_init && audio_pkt_size > 0) {
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long long diff;
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if (packet.pts == AV_NOPTS_VALUE) {
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@ -476,6 +511,7 @@ static void sound_hdaudio_extract_small_block(
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hdaudio->decode_cache_zero
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= hdaudio->decode_cache + diff;
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decode_cache_zero_init = 1;
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}
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while (audio_pkt_size > 0) {
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@ -514,22 +550,18 @@ static void sound_hdaudio_extract_small_block(
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break;
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}
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}
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if (rate_conversion) {
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hdaudio->resample_samples_written
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= audio_resample(hdaudio->resampler,
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hdaudio->resample_cache,
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hdaudio->decode_cache_zero,
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in_frame_size * 7 / 4);
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hdaudio->resample_samples_in =
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in_frame_size * 7 / 4;
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}
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hdaudio->decode_pos = decode_pos;
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if (rate_conversion) {
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sound_hdaudio_run_resampler_seek(hdaudio);
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}
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}
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memcpy(target_buffer, (rate_conversion
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? hdaudio->resample_cache
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: hdaudio->decode_cache_zero) + sample_ofs,
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nb_samples * target_channels * sizeof(short));
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}
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#endif
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@ -575,6 +607,9 @@ void sound_close_hdaudio(struct hdaudio * hdaudio)
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avcodec_close(hdaudio->pCodecCtx);
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av_close_input_file(hdaudio->pFormatCtx);
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MEM_freeN (hdaudio->decode_cache);
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if (hdaudio->resampler) {
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audio_resample_close(hdaudio->resampler);
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}
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if (hdaudio->resample_cache) {
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MEM_freeN(hdaudio->resample_cache);
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}
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@ -564,7 +564,7 @@ static int audio_init(SDL_AudioSpec *desired)
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obtained = (SDL_AudioSpec*)MEM_mallocN(sizeof(SDL_AudioSpec),
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"SDL_AudioSpec");
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audio_initialised = 0;
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desired->callback=audio_fill;
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if ( SDL_OpenAudio(desired, obtained) < 0 ) {
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@ -670,16 +670,18 @@ void audiostream_play(int startframe, uint32_t duration, int mixdown)
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sound_init_audio();
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}
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if (U.mixbufsize && !audio_initialised && !mixdown) {
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if (U.mixbufsize &&
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(!audio_initialised
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|| desired.freq != audio_scene->audio.mixrate
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|| desired.samples != U.mixbufsize)
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&& !mixdown) {
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desired.freq=audio_scene->audio.mixrate;
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desired.format=AUDIO_S16SYS;
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desired.channels=2;
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desired.samples=U.mixbufsize;
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desired.userdata=0;
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if (audio_init(&desired)==0) {
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U.mixbufsize = 0; /* no audio */
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}
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audio_init(&desired);
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}
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audio_startframe = startframe;
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@ -706,7 +708,7 @@ void audiostream_start(int frame)
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void audiostream_scrub(int frame)
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{
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if (U.mixbufsize) audiostream_play(frame, 4096, 0);
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audiostream_play(frame, 4096, 0);
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}
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void audiostream_stop(void)
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@ -721,7 +723,7 @@ int audiostream_pos(void)
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{
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int pos;
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if (U.mixbufsize && audio_scene) {
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if (audio_initialised && audio_scene) {
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pos = (int) (((double)(audio_pos-U.mixbufsize)
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/ ( audio_scene->audio.mixrate*4 ))
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* FPS );
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