New upstream version 24.0.1+dfsg1
This commit is contained in:
Sebastian Ramacher
parent
b14f9eae6d
commit
5a730d6ec3
842 changed files with 42245 additions and 33385 deletions
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@ -23,34 +23,43 @@
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#include <libswresample/swresample.h>
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struct audio_resampler {
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struct SwrContext *context;
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bool opened;
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struct SwrContext *context;
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bool opened;
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uint32_t input_freq;
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uint64_t input_layout;
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uint32_t input_freq;
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uint64_t input_layout;
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enum AVSampleFormat input_format;
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uint8_t *output_buffer[MAX_AV_PLANES];
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uint64_t output_layout;
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uint8_t *output_buffer[MAX_AV_PLANES];
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uint64_t output_layout;
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enum AVSampleFormat output_format;
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int output_size;
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uint32_t output_ch;
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uint32_t output_freq;
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uint32_t output_planes;
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int output_size;
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uint32_t output_ch;
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uint32_t output_freq;
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uint32_t output_planes;
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};
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static inline enum AVSampleFormat convert_audio_format(enum audio_format format)
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{
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switch (format) {
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case AUDIO_FORMAT_UNKNOWN: return AV_SAMPLE_FMT_S16;
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case AUDIO_FORMAT_U8BIT: return AV_SAMPLE_FMT_U8;
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case AUDIO_FORMAT_16BIT: return AV_SAMPLE_FMT_S16;
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case AUDIO_FORMAT_32BIT: return AV_SAMPLE_FMT_S32;
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case AUDIO_FORMAT_FLOAT: return AV_SAMPLE_FMT_FLT;
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case AUDIO_FORMAT_U8BIT_PLANAR: return AV_SAMPLE_FMT_U8P;
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case AUDIO_FORMAT_16BIT_PLANAR: return AV_SAMPLE_FMT_S16P;
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case AUDIO_FORMAT_32BIT_PLANAR: return AV_SAMPLE_FMT_S32P;
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case AUDIO_FORMAT_FLOAT_PLANAR: return AV_SAMPLE_FMT_FLTP;
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case AUDIO_FORMAT_UNKNOWN:
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return AV_SAMPLE_FMT_S16;
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case AUDIO_FORMAT_U8BIT:
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return AV_SAMPLE_FMT_U8;
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case AUDIO_FORMAT_16BIT:
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return AV_SAMPLE_FMT_S16;
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case AUDIO_FORMAT_32BIT:
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return AV_SAMPLE_FMT_S32;
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case AUDIO_FORMAT_FLOAT:
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return AV_SAMPLE_FMT_FLT;
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case AUDIO_FORMAT_U8BIT_PLANAR:
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return AV_SAMPLE_FMT_U8P;
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case AUDIO_FORMAT_16BIT_PLANAR:
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return AV_SAMPLE_FMT_S16P;
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case AUDIO_FORMAT_32BIT_PLANAR:
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return AV_SAMPLE_FMT_S32P;
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case AUDIO_FORMAT_FLOAT_PLANAR:
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return AV_SAMPLE_FMT_FLTP;
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}
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/* shouldn't get here */
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@ -60,14 +69,22 @@ static inline enum AVSampleFormat convert_audio_format(enum audio_format format)
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static inline uint64_t convert_speaker_layout(enum speaker_layout layout)
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{
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switch (layout) {
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case SPEAKERS_UNKNOWN: return 0;
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case SPEAKERS_MONO: return AV_CH_LAYOUT_MONO;
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case SPEAKERS_STEREO: return AV_CH_LAYOUT_STEREO;
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case SPEAKERS_2POINT1: return AV_CH_LAYOUT_SURROUND;
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case SPEAKERS_4POINT0: return AV_CH_LAYOUT_4POINT0;
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case SPEAKERS_4POINT1: return AV_CH_LAYOUT_4POINT1;
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case SPEAKERS_5POINT1: return AV_CH_LAYOUT_5POINT1_BACK;
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case SPEAKERS_7POINT1: return AV_CH_LAYOUT_7POINT1;
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case SPEAKERS_UNKNOWN:
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return 0;
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case SPEAKERS_MONO:
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return AV_CH_LAYOUT_MONO;
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case SPEAKERS_STEREO:
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return AV_CH_LAYOUT_STEREO;
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case SPEAKERS_2POINT1:
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return AV_CH_LAYOUT_SURROUND;
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case SPEAKERS_4POINT0:
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return AV_CH_LAYOUT_4POINT0;
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case SPEAKERS_4POINT1:
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return AV_CH_LAYOUT_4POINT1;
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case SPEAKERS_5POINT1:
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return AV_CH_LAYOUT_5POINT1_BACK;
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case SPEAKERS_7POINT1:
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return AV_CH_LAYOUT_7POINT1;
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}
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/* shouldn't get here */
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@ -75,26 +92,27 @@ static inline uint64_t convert_speaker_layout(enum speaker_layout layout)
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}
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audio_resampler_t *audio_resampler_create(const struct resample_info *dst,
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const struct resample_info *src)
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const struct resample_info *src)
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{
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struct audio_resampler *rs = bzalloc(sizeof(struct audio_resampler));
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int errcode;
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rs->opened = false;
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rs->input_freq = src->samples_per_sec;
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rs->input_layout = convert_speaker_layout(src->speakers);
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rs->input_format = convert_audio_format(src->format);
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rs->output_size = 0;
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rs->output_ch = get_audio_channels(dst->speakers);
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rs->output_freq = dst->samples_per_sec;
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rs->opened = false;
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rs->input_freq = src->samples_per_sec;
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rs->input_layout = convert_speaker_layout(src->speakers);
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rs->input_format = convert_audio_format(src->format);
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rs->output_size = 0;
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rs->output_ch = get_audio_channels(dst->speakers);
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rs->output_freq = dst->samples_per_sec;
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rs->output_layout = convert_speaker_layout(dst->speakers);
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rs->output_format = convert_audio_format(dst->format);
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rs->output_planes = is_audio_planar(dst->format) ? rs->output_ch : 1;
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rs->context = swr_alloc_set_opts(NULL,
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rs->output_layout, rs->output_format, dst->samples_per_sec,
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rs->input_layout, rs->input_format, src->samples_per_sec,
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0, NULL);
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rs->context = swr_alloc_set_opts(NULL, rs->output_layout,
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rs->output_format,
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dst->samples_per_sec, rs->input_layout,
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rs->input_format, src->samples_per_sec,
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0, NULL);
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if (!rs->context) {
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blog(LOG_ERROR, "swr_alloc_set_opts failed");
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@ -104,23 +122,25 @@ audio_resampler_t *audio_resampler_create(const struct resample_info *dst,
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if (rs->input_layout == AV_CH_LAYOUT_MONO && rs->output_ch > 1) {
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const double matrix[MAX_AUDIO_CHANNELS][MAX_AUDIO_CHANNELS] = {
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{1},
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{1, 1},
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{1, 1, 0},
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{1, 1, 1, 1},
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{1, 1, 1, 0, 1},
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{1, 1, 1, 1, 1, 1},
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{1, 1, 1, 0, 1, 1, 1},
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{1, 1, 1, 0, 1, 1, 1, 1},
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{1},
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{1, 1},
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{1, 1, 0},
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{1, 1, 1, 1},
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{1, 1, 1, 0, 1},
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{1, 1, 1, 1, 1, 1},
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{1, 1, 1, 0, 1, 1, 1},
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{1, 1, 1, 0, 1, 1, 1, 1},
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};
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if (swr_set_matrix(rs->context, matrix[rs->output_ch - 1], 1) < 0)
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blog(LOG_DEBUG, "swr_set_matrix failed for mono upmix\n");
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if (swr_set_matrix(rs->context, matrix[rs->output_ch - 1], 1) <
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0)
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blog(LOG_DEBUG,
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"swr_set_matrix failed for mono upmix\n");
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}
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errcode = swr_init(rs->context);
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if (errcode != 0) {
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blog(LOG_ERROR, "avresample_open failed: error code %d",
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errcode);
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errcode);
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audio_resampler_destroy(rs);
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return NULL;
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}
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@ -140,20 +160,21 @@ void audio_resampler_destroy(audio_resampler_t *rs)
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}
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}
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bool audio_resampler_resample(audio_resampler_t *rs,
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uint8_t *output[], uint32_t *out_frames, uint64_t *ts_offset,
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const uint8_t *const input[], uint32_t in_frames)
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bool audio_resampler_resample(audio_resampler_t *rs, uint8_t *output[],
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uint32_t *out_frames, uint64_t *ts_offset,
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const uint8_t *const input[], uint32_t in_frames)
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{
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if (!rs) return false;
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if (!rs)
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return false;
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struct SwrContext *context = rs->context;
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int ret;
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int64_t delay = swr_get_delay(context, rs->input_freq);
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int estimated = (int)av_rescale_rnd(
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delay + (int64_t)in_frames,
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(int64_t)rs->output_freq, (int64_t)rs->input_freq,
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AV_ROUND_UP);
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int estimated = (int)av_rescale_rnd(delay + (int64_t)in_frames,
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(int64_t)rs->output_freq,
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(int64_t)rs->input_freq,
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AV_ROUND_UP);
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*ts_offset = (uint64_t)swr_get_delay(context, 1000000000);
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@ -163,14 +184,13 @@ bool audio_resampler_resample(audio_resampler_t *rs,
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av_freep(&rs->output_buffer[0]);
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av_samples_alloc(rs->output_buffer, NULL, rs->output_ch,
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estimated, rs->output_format, 0);
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estimated, rs->output_format, 0);
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rs->output_size = estimated;
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}
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ret = swr_convert(context,
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rs->output_buffer, rs->output_size,
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(const uint8_t**)input, in_frames);
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ret = swr_convert(context, rs->output_buffer, rs->output_size,
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(const uint8_t **)input, in_frames);
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if (ret < 0) {
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blog(LOG_ERROR, "swr_convert failed: %d", ret);
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