diff options
author | Michael Niedermayer <michaelni@gmx.at> | 2012-01-24 02:41:53 +0100 |
---|---|---|
committer | Michael Niedermayer <michaelni@gmx.at> | 2012-01-24 02:41:53 +0100 |
commit | 0bb57f8bf029427059be21a562527dcfa0e264c9 (patch) | |
tree | 8e6743c4fc1f16f36899bdea87e485735c0d8d59 /libavcodec/aacenc.c | |
parent | b955d4072e3e563b230c9ab4d6575577a3dc7314 (diff) | |
parent | 0fec2cb15cc6ff1fcc724c774ec36abadcb7b6ad (diff) | |
download | ffmpeg-streaming-0bb57f8bf029427059be21a562527dcfa0e264c9.zip ffmpeg-streaming-0bb57f8bf029427059be21a562527dcfa0e264c9.tar.gz |
Merge remote-tracking branch 'qatar/master'
* qatar/master:
Remove ffmpeg.
aacenc: Simplify windowing
aacenc: Move saved overlap samples to the beginning of the same buffer as incoming samples.
aacenc: Deinterleave input samples before processing.
aacenc: Store channel count in AACEncContext.
aacenc: Move Q^3/4 calculation to it's own table
aacenc: Request normalized float samples instead of converting s16 samples to float.
aacpsy: Replace an if with FFMAX in LAME windowing.
aacenc: cosmetics, replace 'rd' with 'bits' in codebook_trellis_rate to make it more clear what is being calculated.
aacpsy: cosmetics, change a FIXME to a NOTE about subshort comparisons
aacenc: cosmetics: move init() and end() to the bottom of the file.
aacenc: aac_encode_init() cleanup
XWD encoder and decoder
vc1: don't read the interpfrm and bfraction elements for interlaced frames
mxfdec: fix memleak on mxf_read_close()
westwood: split the AUD and VQA demuxers into separate files.
Conflicts:
.gitignore
Changelog
Makefile
configure
doc/ffmpeg.texi
ffmpeg.c
libavcodec/Makefile
libavcodec/aacenc.c
libavcodec/allcodecs.c
libavcodec/avcodec.h
libavcodec/version.h
libavformat/Makefile
libavformat/img2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavcodec/aacenc.c')
-rw-r--r-- | libavcodec/aacenc.c | 354 |
1 files changed, 224 insertions, 130 deletions
diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c index 5ab0f1f..a88d75a 100644 --- a/libavcodec/aacenc.c +++ b/libavcodec/aacenc.c @@ -46,6 +46,14 @@ #define AAC_MAX_CHANNELS 6 +#define ERROR_IF(cond, ...) \ + if (cond) { \ + av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \ + return AVERROR(EINVAL); \ + } + +float ff_aac_pow34sf_tab[428]; + static const uint8_t swb_size_1024_96[] = { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44, @@ -136,6 +144,18 @@ static const uint8_t aac_chan_configs[6][5] = { }; /** + * Table to remap channels from Libav's default order to AAC order. + */ +static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = { + { 0 }, + { 0, 1 }, + { 2, 0, 1 }, + { 2, 0, 1, 3 }, + { 2, 0, 1, 3, 4 }, + { 2, 0, 1, 4, 5, 3 }, +}; + +/** * Make AAC audio config object. * @see 1.6.2.1 "Syntax - AudioSpecificConfig" */ @@ -147,7 +167,7 @@ static void put_audio_specific_config(AVCodecContext *avctx) init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8); put_bits(&pb, 5, 2); //object type - AAC-LC put_bits(&pb, 4, s->samplerate_index); //sample rate index - put_bits(&pb, 4, avctx->channels); + put_bits(&pb, 4, s->channels); //GASpecificConfig put_bits(&pb, 1, 0); //frame length - 1024 samples put_bits(&pb, 1, 0); //does not depend on core coder @@ -160,117 +180,80 @@ static void put_audio_specific_config(AVCodecContext *avctx) flush_put_bits(&pb); } -static av_cold int aac_encode_init(AVCodecContext *avctx) -{ - AACEncContext *s = avctx->priv_data; - int i; - const uint8_t *sizes[2]; - uint8_t grouping[AAC_MAX_CHANNELS]; - int lengths[2]; - - avctx->frame_size = 1024; +#define WINDOW_FUNC(type) \ +static void apply_ ##type ##_window(DSPContext *dsp, SingleChannelElement *sce, const float *audio) - for (i = 0; i < 16; i++) - if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i]) - break; - if (i == 16) { - av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate); - return -1; - } - if (avctx->channels > AAC_MAX_CHANNELS) { - av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels); - return -1; - } - if (avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW) { - av_log(avctx, AV_LOG_ERROR, "Unsupported profile %d\n", avctx->profile); - return -1; - } - if (1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * avctx->channels) { - av_log(avctx, AV_LOG_ERROR, "Too many bits per frame requested\n"); - return -1; - } - s->samplerate_index = i; - - dsputil_init(&s->dsp, avctx); - ff_mdct_init(&s->mdct1024, 11, 0, 1.0); - ff_mdct_init(&s->mdct128, 8, 0, 1.0); - // window init - ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024); - ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128); - ff_init_ff_sine_windows(10); - ff_init_ff_sine_windows(7); - - s->chan_map = aac_chan_configs[avctx->channels-1]; - s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0])); - s->cpe = av_mallocz(sizeof(ChannelElement) * s->chan_map[0]); - avctx->extradata = av_mallocz(5 + FF_INPUT_BUFFER_PADDING_SIZE); - avctx->extradata_size = 5; - put_audio_specific_config(avctx); +WINDOW_FUNC(only_long) +{ + const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024; + const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024; + float *out = sce->ret; - sizes[0] = swb_size_1024[i]; - sizes[1] = swb_size_128[i]; - lengths[0] = ff_aac_num_swb_1024[i]; - lengths[1] = ff_aac_num_swb_128[i]; - for (i = 0; i < s->chan_map[0]; i++) - grouping[i] = s->chan_map[i + 1] == TYPE_CPE; - ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], grouping); - s->psypp = ff_psy_preprocess_init(avctx); - s->coder = &ff_aac_coders[s->options.aac_coder]; + dsp->vector_fmul (out, audio, lwindow, 1024); + dsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024); +} - s->lambda = avctx->global_quality ? avctx->global_quality : 120; +WINDOW_FUNC(long_start) +{ + const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024; + const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; + float *out = sce->ret; + + dsp->vector_fmul(out, audio, lwindow, 1024); + memcpy(out + 1024, audio, sizeof(out[0]) * 448); + dsp->vector_fmul_reverse(out + 1024 + 448, audio, swindow, 128); + memset(out + 1024 + 576, 0, sizeof(out[0]) * 448); +} - ff_aac_tableinit(); +WINDOW_FUNC(long_stop) +{ + const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024; + const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128; + float *out = sce->ret; + + memset(out, 0, sizeof(out[0]) * 448); + dsp->vector_fmul(out + 448, audio + 448, swindow, 128); + memcpy(out + 576, audio + 576, sizeof(out[0]) * 448); + dsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024); +} - return 0; +WINDOW_FUNC(eight_short) +{ + const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; + const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128; + const float *in = audio + 448; + float *out = sce->ret; + + for (int w = 0; w < 8; w++) { + dsp->vector_fmul (out, in, w ? pwindow : swindow, 128); + out += 128; + in += 128; + dsp->vector_fmul_reverse(out, in, swindow, 128); + out += 128; + } } -static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s, - SingleChannelElement *sce, short *audio) +static void (*const apply_window[4])(DSPContext *dsp, SingleChannelElement *sce, const float *audio) = { + [ONLY_LONG_SEQUENCE] = apply_only_long_window, + [LONG_START_SEQUENCE] = apply_long_start_window, + [EIGHT_SHORT_SEQUENCE] = apply_eight_short_window, + [LONG_STOP_SEQUENCE] = apply_long_stop_window +}; + +static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce, + float *audio) { - int i, k; - const int chans = avctx->channels; - const float * lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024; - const float * swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; - const float * pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128; + int i; float *output = sce->ret; - if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) { - memcpy(output, sce->saved, sizeof(float)*1024); - if (sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE) { - memset(output, 0, sizeof(output[0]) * 448); - for (i = 448; i < 576; i++) - output[i] = sce->saved[i] * pwindow[i - 448]; - for (i = 576; i < 704; i++) - output[i] = sce->saved[i]; - } - if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) { - for (i = 0; i < 1024; i++) { - output[i+1024] = audio[i * chans] * lwindow[1024 - i - 1]; - sce->saved[i] = audio[i * chans] * lwindow[i]; - } - } else { - for (i = 0; i < 448; i++) - output[i+1024] = audio[i * chans]; - for (; i < 576; i++) - output[i+1024] = audio[i * chans] * swindow[576 - i - 1]; - memset(output+1024+576, 0, sizeof(output[0]) * 448); - for (i = 0; i < 1024; i++) - sce->saved[i] = audio[i * chans]; - } + apply_window[sce->ics.window_sequence[0]](&s->dsp, sce, audio); + + if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output); - } else { - for (k = 0; k < 1024; k += 128) { - for (i = 448 + k; i < 448 + k + 256; i++) - output[i - 448 - k] = (i < 1024) - ? sce->saved[i] - : audio[(i-1024)*chans]; - s->dsp.vector_fmul (output, output, k ? swindow : pwindow, 128); - s->dsp.vector_fmul_reverse(output+128, output+128, swindow, 128); - s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + k, output); - } - for (i = 0; i < 1024; i++) - sce->saved[i] = audio[i * chans]; - } + else + for (i = 0; i < 1024; i += 128) + s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + i, output + i*2); + memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024); } /** @@ -488,11 +471,37 @@ static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s, put_bits(&s->pb, 12 - padbits, 0); } +/* + * Deinterleave input samples. + * Channels are reordered from Libav's default order to AAC order. + */ +static void deinterleave_input_samples(AACEncContext *s, + const float *samples) +{ + int ch, i; + const int sinc = s->channels; + const uint8_t *channel_map = aac_chan_maps[sinc - 1]; + + /* deinterleave and remap input samples */ + for (ch = 0; ch < sinc; ch++) { + const float *sptr = samples + channel_map[ch]; + + /* copy last 1024 samples of previous frame to the start of the current frame */ + memcpy(&s->planar_samples[ch][0], &s->planar_samples[ch][1024], 1024 * sizeof(s->planar_samples[0][0])); + + /* deinterleave */ + for (i = 1024; i < 1024 * 2; i++) { + s->planar_samples[ch][i] = *sptr; + sptr += sinc; + } + } +} + static int aac_encode_frame(AVCodecContext *avctx, uint8_t *frame, int buf_size, void *data) { AACEncContext *s = avctx->priv_data; - int16_t *samples = s->samples, *samples2, *la; + float **samples = s->planar_samples, *samples2, *la, *overlap; ChannelElement *cpe; int i, ch, w, g, chans, tag, start_ch; int chan_el_counter[4]; @@ -500,27 +509,15 @@ static int aac_encode_frame(AVCodecContext *avctx, if (s->last_frame) return 0; + if (data) { - if (!s->psypp) { - memcpy(s->samples + 1024 * avctx->channels, data, - 1024 * avctx->channels * sizeof(s->samples[0])); - } else { - start_ch = 0; - samples2 = s->samples + 1024 * avctx->channels; - for (i = 0; i < s->chan_map[0]; i++) { - tag = s->chan_map[i+1]; - chans = tag == TYPE_CPE ? 2 : 1; - ff_psy_preprocess(s->psypp, (uint16_t*)data + start_ch, - samples2 + start_ch, start_ch, chans); - start_ch += chans; - } - } + deinterleave_input_samples(s, data); + if (s->psypp) + ff_psy_preprocess(s->psypp, s->planar_samples, s->channels); } - if (!avctx->frame_number) { - memcpy(s->samples, s->samples + 1024 * avctx->channels, - 1024 * avctx->channels * sizeof(s->samples[0])); + + if (!avctx->frame_number) return 0; - } start_ch = 0; for (i = 0; i < s->chan_map[0]; i++) { @@ -531,8 +528,9 @@ static int aac_encode_frame(AVCodecContext *avctx, for (ch = 0; ch < chans; ch++) { IndividualChannelStream *ics = &cpe->ch[ch].ics; int cur_channel = start_ch + ch; - samples2 = samples + cur_channel; - la = samples2 + (448+64) * avctx->channels; + overlap = &samples[cur_channel][0]; + samples2 = overlap + 1024; + la = samples2 + (448+64); if (!data) la = NULL; if (tag == TYPE_LFE) { @@ -560,7 +558,7 @@ static int aac_encode_frame(AVCodecContext *avctx, for (w = 0; w < ics->num_windows; w++) ics->group_len[w] = wi[ch].grouping[w]; - apply_window_and_mdct(avctx, s, &cpe->ch[ch], samples2); + apply_window_and_mdct(s, &cpe->ch[ch], overlap); } start_ch += chans; } @@ -626,8 +624,8 @@ static int aac_encode_frame(AVCodecContext *avctx, } frame_bits = put_bits_count(&s->pb); - if (frame_bits <= 6144 * avctx->channels - 3) { - s->psy.bitres.bits = frame_bits / avctx->channels; + if (frame_bits <= 6144 * s->channels - 3) { + s->psy.bitres.bits = frame_bits / s->channels; break; } @@ -648,8 +646,7 @@ static int aac_encode_frame(AVCodecContext *avctx, if (!data) s->last_frame = 1; - memcpy(s->samples, s->samples + 1024 * avctx->channels, - 1024 * avctx->channels * sizeof(s->samples[0])); + return put_bits_count(&s->pb)>>3; } @@ -660,12 +657,109 @@ static av_cold int aac_encode_end(AVCodecContext *avctx) ff_mdct_end(&s->mdct1024); ff_mdct_end(&s->mdct128); ff_psy_end(&s->psy); - ff_psy_preprocess_end(s->psypp); - av_freep(&s->samples); + if (s->psypp) + ff_psy_preprocess_end(s->psypp); + av_freep(&s->buffer.samples); av_freep(&s->cpe); return 0; } +static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s) +{ + int ret = 0; + + dsputil_init(&s->dsp, avctx); + + // window init + ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024); + ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128); + ff_init_ff_sine_windows(10); + ff_init_ff_sine_windows(7); + + if (ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0)) + return ret; + if (ret = ff_mdct_init(&s->mdct128, 8, 0, 32768.0)) + return ret; + + return 0; +} + +static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s) +{ + FF_ALLOCZ_OR_GOTO(avctx, s->buffer.samples, 3 * 1024 * s->channels * sizeof(s->buffer.samples[0]), alloc_fail); + FF_ALLOCZ_OR_GOTO(avctx, s->cpe, sizeof(ChannelElement) * s->chan_map[0], alloc_fail); + FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail); + + for(int ch = 0; ch < s->channels; ch++) + s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch; + + return 0; +alloc_fail: + return AVERROR(ENOMEM); +} + +static av_cold int aac_encode_init(AVCodecContext *avctx) +{ + AACEncContext *s = avctx->priv_data; + int i, ret = 0; + const uint8_t *sizes[2]; + uint8_t grouping[AAC_MAX_CHANNELS]; + int lengths[2]; + + avctx->frame_size = 1024; + + for (i = 0; i < 16; i++) + if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i]) + break; + + s->channels = avctx->channels; + + ERROR_IF(i == 16, + "Unsupported sample rate %d\n", avctx->sample_rate); + ERROR_IF(s->channels > AAC_MAX_CHANNELS, + "Unsupported number of channels: %d\n", s->channels); + ERROR_IF(avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW, + "Unsupported profile %d\n", avctx->profile); + ERROR_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels, + "Too many bits per frame requested\n"); + + s->samplerate_index = i; + + s->chan_map = aac_chan_configs[s->channels-1]; + + if (ret = dsp_init(avctx, s)) + goto fail; + + if (ret = alloc_buffers(avctx, s)) + goto fail; + + avctx->extradata_size = 5; + put_audio_specific_config(avctx); + + sizes[0] = swb_size_1024[i]; + sizes[1] = swb_size_128[i]; + lengths[0] = ff_aac_num_swb_1024[i]; + lengths[1] = ff_aac_num_swb_128[i]; + for (i = 0; i < s->chan_map[0]; i++) + grouping[i] = s->chan_map[i + 1] == TYPE_CPE; + if (ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], grouping)) + goto fail; + s->psypp = ff_psy_preprocess_init(avctx); + s->coder = &ff_aac_coders[s->options.aac_coder]; + + s->lambda = avctx->global_quality ? avctx->global_quality : 120; + + ff_aac_tableinit(); + + for (i = 0; i < 428; i++) + ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i])); + + return 0; +fail: + aac_encode_end(avctx); + return ret; +} + #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM static const AVOption aacenc_options[] = { {"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.dbl = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"}, @@ -692,7 +786,7 @@ AVCodec ff_aac_encoder = { .encode = aac_encode_frame, .close = aac_encode_end, .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL, - .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE}, + .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE}, .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"), .priv_class = &aacenc_class, }; |