[FFmpeg-soc] [soc]: r2615 - in aac: aac.c aactab.h
superdump
subversion at mplayerhq.hu
Tue Jul 1 12:12:10 CEST 2008
Author: superdump
Date: Tue Jul 1 12:12:10 2008
New Revision: 2615
Log:
Punctuation, spelling, grammar, etcetera
Modified:
aac/aac.c
aac/aactab.h
Modified: aac/aac.c
==============================================================================
--- aac/aac.c (original)
+++ aac/aac.c Tue Jul 1 12:12:10 2008
@@ -127,7 +127,7 @@ enum {
};
/**
- * Window sequences
+ * window sequences
*/
enum WindowSequence {
ONLY_LONG_SEQUENCE,
@@ -137,7 +137,7 @@ enum WindowSequence {
};
/**
- * Special codebooks
+ * special codebooks
*/
enum {
ZERO_HCB = 0,
@@ -152,7 +152,7 @@ enum {
#define IS_CODEBOOK_UNSIGNED(x) ((x - 1) & 10)
/**
- * Channel types
+ * channel types
*/
enum {
AAC_CHANNEL_FRONT = 1,
@@ -163,7 +163,7 @@ enum {
};
/**
- * Mix-down channel types
+ * mix-down channel types
*/
enum {
MIXDOWN_CENTER,
@@ -172,18 +172,16 @@ enum {
};
/**
- * Program configuration - describes how channels are arranged
- *
- * Either read from stream (ID_PCE) or created based on a default
- * fixed channel arrangement.
+ * Program configuration - describes how channels are arranged. Either read from
+ * stream (ID_PCE) or created based on a default fixed channel arrangement.
*/
typedef struct {
- int che_type[4][MAX_TAGID]; ///< Channel element type with the first index as the first 4 raw_data_block IDs
+ int che_type[4][MAX_TAGID]; ///< channel element type with the first index as the first 4 raw_data_block IDs
- int mono_mixdown; ///< The SCE tag to use if user requests mono output, -1 if not available
- int stereo_mixdown; ///< The CPE tag to use if user requests stereo output, -1 if not available
+ int mono_mixdown; ///< The SCE tag to use if user requests mono output, -1 if not available.
+ int stereo_mixdown; ///< The CPE tag to use if user requests stereo output, -1 if not available.
int mixdown_coeff_index; ///< 0-3
- int pseudo_surround; ///< Mix surround channels out of phase
+ int pseudo_surround; ///< Mix surround channels out of phase.
} ProgramConfig;
@@ -207,7 +205,7 @@ typedef struct {
int max_sfb; ///< number of scalefactor bands per group
enum WindowSequence window_sequence;
enum WindowSequence window_sequence_prev;
- uint8_t use_kb_window[2]; ///< If set, use Kaiser-Bessel window, otherwise use a sinus window
+ uint8_t use_kb_window[2]; ///< If set, use Kaiser-Bessel window, otherwise use a sinus window.
int num_window_groups;
uint8_t group_len[8];
LongTermPrediction ltp;
@@ -240,9 +238,7 @@ typedef struct {
} MidSideStereo;
/**
- * Dynamic Range Control
- *
- * DRC is decoded from the bitstream but not further processed
+ * Dynamic Range Control - decoded from the bitstream but not processed further.
*/
typedef struct {
int pce_instance_tag;
@@ -259,7 +255,7 @@ typedef struct {
} DynamicRangeControl;
/**
- * Pulse tool
+ * pulse tool
*/
typedef struct {
int present;
@@ -270,7 +266,7 @@ typedef struct {
} Pulse;
/**
- * Parameters for the SSR Inverse Polyphase Quadrature Filter
+ * parameters for the SSR Inverse Polyphase Quadrature Filter
*/
typedef struct {
float q[4][4];
@@ -279,7 +275,7 @@ typedef struct {
} ssr_context;
/**
- * Per-element gain control for SSR
+ * per-element gain control for SSR
*/
typedef struct {
int max_band;
@@ -290,23 +286,22 @@ typedef struct {
} ScalableSamplingRate;
/**
- * Coupling parameters
+ * coupling parameters
*/
typedef struct {
- int ind_sw; ///< Set if independent coupling (i.e. after IMDCT)
- int domain; ///< Controls if coupling is performed before (0) or after (1) the TNS decoding of the target channels
- int num_coupled; ///< Number of target elements
- int is_cpe[9]; ///< Set if target is an CPE (otherwise it's an SCE)
- int tag_select[9]; ///< Element tag index
- int l[9]; ///< Apply gain to left channel of an CPE
- int r[9]; ///< Apply gain to right channel of an CPE
+ int ind_sw; ///< Set if independent coupling (i.e. after IMDCT).
+ int domain; ///< Controls if coupling is performed before (0) or after (1) the TNS decoding of the target channels.
+ int num_coupled; ///< number of target elements
+ int is_cpe[9]; ///< Set if target is an CPE (otherwise it's an SCE).
+ int tag_select[9]; ///< element tag index
+ int l[9]; ///< Apply gain to left channel of a CPE.
+ int r[9]; ///< Apply gain to right channel of a CPE.
float gain[18][8][64];
} ChannelCoupling;
/**
- * Single Channel Element
- * Used for both SCE and LFE elements
+ * Single Channel Element - used for both SCE and LFE elements.
*/
typedef struct {
float mixing_gain; /**< Channel gain (not used by AAC bitstream).
@@ -326,28 +321,27 @@ typedef struct {
} SingleChannelElement;
/**
- * Channel element
- * Generic struct for SCE/CPE/CCE/LFE
+ * channel element - generic struct for SCE/CPE/CCE/LFE
*/
typedef struct {
// CPE specific
- int common_window; ///< Set if channels share a common 'IndividualChannelStream' in bitstream
+ int common_window; ///< Set if channels share a common 'IndividualChannelStream' in bitstream.
MidSideStereo ms;
- // Shared
+ // shared
SingleChannelElement ch[2];
// CCE specific
ChannelCoupling coup;
} ChannelElement;
/**
- * Main AAC context
+ * main AAC context
*/
typedef struct {
AVCodecContext * avccontext;
MPEG4AudioConfig m4ac;
- int is_saved; ///< Set if elements have stored overlap from previous frame
+ int is_saved; ///< Set if elements have stored overlap from previous frame.
DynamicRangeControl che_drc;
/**
@@ -359,7 +353,7 @@ typedef struct {
/** @} */
/**
- * @defgroup temporary Aligned temporary buffers (we do not want to have these on stack)
+ * @defgroup temporary aligned temporary buffers (We do not want to have these on the stack.)
* @{
*/
DECLARE_ALIGNED_16(float, buf_mdct[2048]);
@@ -367,7 +361,7 @@ typedef struct {
/** @} */
/**
- * @defgroup tables Computed / setup during initialization
+ * @defgroup tables Computed / set up during initialization.
* @{
*/
MDCTContext mdct;
@@ -379,15 +373,15 @@ typedef struct {
/** @} */
/**
- * @defgroup output Members used for output interleaving and down-mixing
+ * @defgroup output Members used for output interleaving and down-mixing.
* @{
*/
- float* interleaved_output; ///< Interim buffer for interleaving PCM samples
- float *output_data[MAX_CHANNELS]; ///< Points to each element's 'ret' buffer (PCM output)
- ChannelElement *mm[3]; ///< Center/Front/Back channel elements to use for matrix mix-down
- float add_bias; ///< Offset for dsp.float_to_int16
- float sf_scale; ///< Pre-scale for correct IMDCT and dsp.float_to_int16
- int sf_offset; ///< Offset into pow2sf_tab as appropriate for dsp.float_to_int16
+ float* interleaved_output; ///< Interim buffer for interleaving PCM samples.
+ float *output_data[MAX_CHANNELS]; ///< Points to each element's 'ret' buffer (PCM output).
+ ChannelElement *mm[3]; ///< Center/Front/Back channel elements to use for matrix mix-down.
+ float add_bias; ///< offset for dsp.float_to_int16
+ float sf_scale; ///< Pre-scale for correct IMDCT and dsp.float_to_int16.
+ int sf_offset; ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16
/** @} */
} AACContext;
@@ -437,7 +431,7 @@ static void ssr_context_init(ssr_context
#endif /* AAC_SSR */
/**
- * Free a Channel Element
+ * Free a channel element.
*/
static void che_freep(ChannelElement **s) {
if(!*s)
@@ -451,9 +445,9 @@ static void che_freep(ChannelElement **s
/**
* Configure output channel order and optional mixing based on the current
- * program configuration element and user requested channels
+ * program configuration element and user requested channels.
*
- * \param newpcs New program configuration struct - we only do something if it differs from the current one
+ * \param newpcs New program configuration struct - we only do something if it differs from the current one.
*/
static int output_configure(AACContext *ac, ProgramConfig *newpcs) {
AVCodecContext *avctx = ac->avccontext;
@@ -463,7 +457,7 @@ static int output_configure(AACContext *
ChannelElement *mixdown[3] = { NULL, NULL, NULL };
static const float mixdowncoeff[4] = {
- /* Matrix mix-down coefficient, Table 4.70 */
+ /* matrix mix-down coefficient, table 4.70 */
1. / M_SQRT2,
1. / 2.,
1. / (2 * M_SQRT2),
@@ -476,14 +470,14 @@ static int output_configure(AACContext *
*pcs = *newpcs;
/* Allocate or free elements depending on if they are in the
- current program configuration struct */
-
- /* Set up default 1:1 output mapping
+ * current program configuration struct.
+ *
+ * Set up default 1:1 output mapping.
*
* For a 5.1 stream the output order will be:
* [ Front Left ] [ Front Right ] [ Center ] [ LFE ] [ Surround Left ] [ Surround Right ]
*
- * While at it: locate front, center and back for matrix mix-down further down
+ * Locate front, center and back channels for later matrix mix-down.
*/
for(i = 0; i < MAX_TAGID; i++) {
@@ -511,7 +505,7 @@ static int output_configure(AACContext *
ac->mm[MIXDOWN_FRONT] = ac->mm[MIXDOWN_BACK] = ac->mm[MIXDOWN_CENTER] = NULL;
- /* Check for matrix mix-down to mono or stereo */
+ /* Check for matrix mix-down to mono or stereo. */
if(avctx->request_channels && avctx->request_channels <= 2 &&
avctx->request_channels != channels) {
@@ -519,13 +513,13 @@ static int output_configure(AACContext *
if((avctx->request_channels == 1 && pcs->mono_mixdown != -1) ||
(avctx->request_channels == 2 && pcs->stereo_mixdown != -1)) {
/* Add support for this as soon as we get a sample so we can figure out
- exactly how this is supposed to work */
+ exactly how this is supposed to work. */
av_log(avctx, AV_LOG_ERROR,
"Mix-down using pre-mixed elements is not supported, please file a bug. "
- "Reverting to matrix mix-down\n");
+ "Reverting to matrix mix-down.\n");
}
- /* We need 'center + L + R + sL + sR' for matrix mix-down */
+ /* We need 'center + L + R + sL + sR' for matrix mix-down. */
if(mixdown[MIXDOWN_CENTER] && mixdown[MIXDOWN_FRONT] && mixdown[MIXDOWN_BACK]) {
a = mixdowncoeff[pcs->mixdown_coeff_index];
@@ -544,7 +538,7 @@ static int output_configure(AACContext *
channels = avctx->request_channels;
} else {
- av_log(avctx, AV_LOG_WARNING, "Matrix mixing from %d to %d channels in not supported\n",
+ av_log(avctx, AV_LOG_WARNING, "Matrix mixing from %d to %d channels is not supported.\n",
channels, avctx->request_channels);
}
}
@@ -556,11 +550,11 @@ static int output_configure(AACContext *
/**
- * Decode an array of 4 bit tag IDs, optionally interleaved with a stereo/mono switching bit
+ * Decode an array of 4 bit tag IDs, optionally interleaved with a stereo/mono switching bit.
*
- * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present
- * @param sce_map Mono (Single Channel Element) map
- * @param type Speaker type/position for these channels
+ * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present.
+ * @param sce_map mono (Single Channel Element) map
+ * @param type speaker type/position for these channels
*/
static void program_config_element_parse_tags(GetBitContext * gb, int *cpe_map,
int *sce_map, int n, int type) {
@@ -573,8 +567,7 @@ static void program_config_element_parse
/**
- * Parse program configuration element
- * reference: Table 4.2
+ * Parse program configuration element; reference: table 4.2.
*/
static int program_config_element(AACContext * ac, GetBitContext * gb) {
ProgramConfig pcs;
@@ -586,7 +579,7 @@ static int program_config_element(AACCon
ac->m4ac.sampling_index = get_bits(gb, 4);
if(ac->m4ac.sampling_index > 12) {
- av_log(ac->avccontext, AV_LOG_ERROR, "Invalid sampling rate index %d\n", ac->m4ac.sampling_index);
+ av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
return -1;
}
ac->m4ac.sample_rate = ff_mpeg4audio_sample_rates[ac->m4ac.sampling_index];
@@ -623,7 +616,7 @@ static int program_config_element(AACCon
/**
* Set up ProgramConfig, but based on a default channel configuration
- * as specified in Table 1.17
+ * as specified in table 1.17.
*/
static int program_config_element_default(AACContext *ac, int channels)
{
@@ -631,54 +624,53 @@ static int program_config_element_defaul
memset(&pcs, 0, sizeof(ProgramConfig));
- /* Pre-mixed down-mix outputs are not available */
+ /* Pre-mixed down-mix outputs are not available. */
pcs.mono_mixdown = -1;
pcs.stereo_mixdown = -1;
if(channels < 1 || channels > 7) {
- av_log(ac->avccontext, AV_LOG_ERROR, "Invalid default channel configuration (%d channels)\n",
+ av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d channels)\n",
channels);
return -1;
}
- /* Default channel configurations:
+ /* default channel configurations:
*
- * 1ch : Front Center (Mono)
- * 2ch : L + R (Stereo)
- * 3ch : Front Center + L + R
- * 4ch : Front Center + L + R + Back Center
- * 5ch : Front Center + L + R + Back Stereo
- * 6ch : Front Center + L + R + Back Stereo + LFE
- * 7ch : Front Center + L + R + Outer Front Left + Outer Front Right + Back Stereo + LFE
+ * 1ch : front center (mono)
+ * 2ch : L + R (stereo)
+ * 3ch : front center + L + R
+ * 4ch : front center + L + R + back center
+ * 5ch : front center + L + R + back stereo
+ * 6ch : front center + L + R + back stereo + LFE
+ * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE
*/
if(channels != 2)
- pcs.che_type[ID_SCE][0] = AAC_CHANNEL_FRONT; // Front Center (or Mono)
+ pcs.che_type[ID_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono)
if(channels > 1)
- pcs.che_type[ID_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or Stereo)
+ pcs.che_type[ID_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo)
if(channels == 4)
- pcs.che_type[ID_SCE][1] = AAC_CHANNEL_BACK; // Back Center
+ pcs.che_type[ID_SCE][1] = AAC_CHANNEL_BACK; // back center
if(channels > 4)
pcs.che_type[ID_CPE][(channels == 7) + 1]
- = AAC_CHANNEL_BACK; // Back Stereo
+ = AAC_CHANNEL_BACK; // back stereo
if(channels > 5)
pcs.che_type[ID_LFE][0] = AAC_CHANNEL_LFE; // LFE
if(channels == 7)
- pcs.che_type[ID_CPE][1] = AAC_CHANNEL_FRONT; // Outer Front Left + Outer Front Right
+ pcs.che_type[ID_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right
return output_configure(ac, &pcs);
}
/**
- * Parse GA "General Audio" specific configuration
- * reference: Table 4.1
+ * Parse GA "General Audio" specific configuration; reference: table 4.1.
*/
static int GASpecificConfig(AACContext * ac, GetBitContext * gb, int channels) {
int ext;
if(get_bits1(gb)) { // frameLengthFlag
- av_log(ac->avccontext, AV_LOG_ERROR, "960/120 MDCT window is not supported\n");
+ av_log(ac->avccontext, AV_LOG_ERROR, "960/120 MDCT window is not supported.\n");
return -1;
}
@@ -722,8 +714,7 @@ static int GASpecificConfig(AACContext *
/**
- * Parse audio specific configuration
- * reference: Table 1.13
+ * Parse audio specific configuration; reference: table 1.13.
*/
static int AudioSpecificConfig(AACContext * ac, void *data, int data_size) {
GetBitContext gb;
@@ -748,7 +739,7 @@ static int AudioSpecificConfig(AACContex
return -1;
break;
default:
- av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported\n",
+ av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type);
return -1;
}
@@ -831,12 +822,12 @@ static int aac_decode_init(AVCodecContex
dsputil_init(&ac->dsp, avccontext);
- /* Initialize RNG dither */
+ /* Initialize RNG dither. */
av_init_random(0x1f2e3d4c, &ac->random_state);
- // -1024 - compensate wrong IMDCT method
- // 32768 - values in AAC build for ready float->int 16 bit audio, using
- // BIAS method instead needs values -1<x<1
+ // -1024 - Compensate wrong IMDCT method.
+ // 32768 - Required to scale values to the correct range for the bias method
+ // for float to int16 conversion.
if(ac->dsp.float_to_int16 == ff_float_to_int16_c) {
ac->add_bias = 385.0f;
@@ -885,11 +876,10 @@ static int aac_decode_init(AVCodecContex
return 0;
}
-// Parsers implementation
+// parser implementation
/**
- * Decode a data_stream_element
- * reference: Table 4.10
+ * Decode a data_stream_element; reference: table 4.10.
*/
static void data_stream_element(AACContext * ac, GetBitContext * gb, int id) {
int byte_align = get_bits1(gb);
@@ -916,13 +906,12 @@ static void decode_ltp_data(AACContext *
#endif /* AAC_LTP */
/**
- * Decode Individual Channel Stream info
- * reference: table 4.6
+ * Decode Individual Channel Stream info; reference: table 4.6.
*/
static int decode_ics_info(AACContext * ac, GetBitContext * gb, int common_window, IndividualChannelStream * ics) {
uint8_t grouping;
if (get_bits1(gb)) {
- av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set\n");
+ av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n");
return -1;
}
ics->window_sequence_prev = ics->window_sequence;
@@ -947,7 +936,7 @@ static int decode_ics_info(AACContext *
ics->num_swb = num_swb_128[ac->m4ac.sampling_index];
if(ics->max_sfb > ics->num_swb) {
av_log(ac->avccontext, AV_LOG_ERROR,
- "Number of scalefactor bands in group (%d) exceeds limit (%d)\n",
+ "Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
ics->max_sfb, ics->num_swb);
return -1;
}
@@ -960,7 +949,7 @@ static int decode_ics_info(AACContext *
ics->num_swb = num_swb_1024[ac->m4ac.sampling_index];
if(ics->max_sfb > ics->num_swb) {
av_log(ac->avccontext, AV_LOG_ERROR,
- "Number of scalefactor bands in group (%d) exceeds limit (%d)\n",
+ "Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
ics->max_sfb, ics->num_swb);
return -1;
}
@@ -983,7 +972,7 @@ static int decode_ics_info(AACContext *
}
#else /* AAC_LTP */
av_log(ac->avccontext, AV_LOG_ERROR,
- "Predictor bit set but LTP is not supported\n");
+ "Predictor bit set but LTP is not supported.\n");
return -1;
#endif /* AAC_LTP */
} else {
@@ -1002,8 +991,7 @@ static inline float ivquant(AACContext *
}
/**
- * Decode section_data payload
- * reference: Table 4.46
+ * Decode section_data payload; reference: table 4.46.
*/
static int decode_section_data(AACContext * ac, GetBitContext * gb, IndividualChannelStream * ics, int cb[][64], int cb_run_end[][64]) {
int g;
@@ -1015,7 +1003,7 @@ static int decode_section_data(AACContex
int sect_len_incr;
int sect_cb = get_bits(gb, 4);
if (sect_cb == 12) {
- av_log(ac->avccontext, AV_LOG_ERROR, "Invalid codebook\n");
+ av_log(ac->avccontext, AV_LOG_ERROR, "invalid codebook\n");
return -1;
}
while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits)-1)
@@ -1023,7 +1011,7 @@ static int decode_section_data(AACContex
sect_len += sect_len_incr;
if (sect_len > ics->max_sfb) {
av_log(ac->avccontext, AV_LOG_ERROR,
- "Number of codebooks (%d) exceeds limit (%d)\n",
+ "Number of codebooks (%d) exceeds limit (%d).\n",
sect_len, ics->max_sfb);
return -1;
}
@@ -1037,8 +1025,7 @@ static int decode_section_data(AACContex
}
/**
- * Decode scale_factor_data
- * reference: Table 4.47
+ * Decode scale_factor_data; reference: table 4.47.
*/
static int decode_scale_factor_data(AACContext * ac, GetBitContext * gb, float mix_gain, unsigned int global_gain, IndividualChannelStream * ics, const int cb[][64], const int cb_run_end[][64], float sf[][64]) {
const int sf_offset = ac->sf_offset + (ics->window_sequence == EIGHT_SHORT_SEQUENCE ? 12 : 0);
@@ -1061,7 +1048,7 @@ static int decode_scale_factor_data(AACC
offset[2] += get_vlc2(gb, mainvlc.table, 7, 3) - 60;
if(offset[2] > 255) {
av_log(ac->avccontext, AV_LOG_ERROR,
- "%s (%d) out of range", sf_str[2], offset[2]);
+ "%s (%d) out of range.\n", sf_str[2], offset[2]);
return -1;
}
sf[g][i] = pow2sf_tab[-offset[2] + 300];
@@ -1076,7 +1063,7 @@ static int decode_scale_factor_data(AACC
offset[1] += get_vlc2(gb, mainvlc.table, 7, 3) - 60;
if(offset[1] > 255) {
av_log(ac->avccontext, AV_LOG_ERROR,
- "%s (%d) out of range", sf_str[1], offset[1]);
+ "%s (%d) out of range.\n", sf_str[1], offset[1]);
return -1;
}
sf[g][i] = -pow2sf_tab[ offset[1] + sf_offset];
@@ -1088,7 +1075,7 @@ static int decode_scale_factor_data(AACC
offset[0] += get_vlc2(gb, mainvlc.table, 7, 3) - 60;
if(offset[0] > 255) {
av_log(ac->avccontext, AV_LOG_ERROR,
- "%s (%d) out of range", sf_str[0], offset[0]);
+ "%s (%d) out of range.\n", sf_str[0], offset[0]);
return -1;
}
sf[g][i] = -pow2sf_tab[ offset[0] + sf_offset];
@@ -1180,8 +1167,7 @@ static void decode_ms_data(AACContext *
}
/**
- * Decode spectral data
- * reference: Table 4.50
+ * Decode spectral data; reference: table 4.50.
*/
static int decode_spectral_data(AACContext * ac, GetBitContext * gb, const IndividualChannelStream * ics, const int cb[][64], int icoef[1024]) {
int i, k, g;
@@ -1214,11 +1200,11 @@ static int decode_spectral_data(AACConte
for (j = 0; j < 2; j++) {
if (vq_ptr[j] == 16) {
int n = 4;
- /* Total length of escape_sequence must be < 22 bits according to spec */
- /* i.e. max is 11111111110xxxxxxxxxx */
+ /* The total length of escape_sequence must be < 22 bits according
+ to the specification (i.e. max is 11111111110xxxxxxxxxx). */
while (get_bits1(gb) && n < 15) n++;
if(n == 15) {
- av_log(ac->avccontext, AV_LOG_ERROR, "Error in spectral data, ESC overflow\n");
+ av_log(ac->avccontext, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
return -1;
}
icoef[coef_idx + j] *= (1<<n) + get_bits(gb, n);
@@ -1278,8 +1264,7 @@ static void quant_to_spec_tool(AACContex
}
/**
- * Decode an individual_channel_stream payload
- * reference: Table 4.44
+ * Decode an individual_channel_stream payload; reference: table 4.44.
*/
static int decode_ics(AACContext * ac, GetBitContext * gb, int common_window, int scale_flag, SingleChannelElement * sce) {
int icoeffs[1024];
@@ -1305,7 +1290,7 @@ static int decode_ics(AACContext * ac, G
if (!scale_flag) {
if ((pulse.present = get_bits1(gb))) {
if (ics->window_sequence == EIGHT_SHORT_SEQUENCE) {
- av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence\n");
+ av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
return -1;
}
decode_pulse_data(ac, gb, &pulse);
@@ -1316,7 +1301,7 @@ static int decode_ics(AACContext * ac, G
#ifdef AAC_SSR
if (decode_gain_control_data(ac, gb, sce)) return -1;
#else
- av_log(ac->avccontext, AV_LOG_ERROR, "SSR not supported\n");
+ av_log(ac->avccontext, AV_LOG_ERROR, "SSR not supported.\n");
return -1;
#endif
}
@@ -1385,8 +1370,7 @@ static void intensity_tool(AACContext *
}
/**
- * Decode a channel_pair_element
- * reference: Table 4.4
+ * Decode a channel_pair_element; reference: table 4.4.
*/
static int decode_cpe(AACContext * ac, GetBitContext * gb, int id) {
int i;
@@ -1557,7 +1541,7 @@ static int dynamic_range_info(AACContext
return n;
}
-/** Parse extension data (incomplete)
+/** Parse extension data (incomplete).
*/
static int extension_payload(AACContext * ac, GetBitContext * gb, int cnt) {
int i = 0;
@@ -1674,7 +1658,7 @@ static void window_ltp_tool(AACContext *
ac->dsp.vector_fmul_reverse(buf + 1024 + 448, in + 1024 + 448, swindow, 128);
memset(buf + 1024 + 576, 0, 448 * sizeof(float));
}
- ff_mdct_calc(ac->mdct_ltp, out, buf, in); // using in as buffer for mdct
+ ff_mdct_calc(ac->mdct_ltp, out, buf, in); // Using in as buffer for MDCT.
}
static void ltp_trans(AACContext * ac, SingleChannelElement * sce) {
@@ -1702,7 +1686,7 @@ static void ltp_trans(AACContext * ac, S
/**
- * @todo: Replace this with float_to_int16()
+ * @todo: Replace this with float_to_int16().
*/
static inline int16_t ltp_round(float x) {
if (x >= 0)
@@ -1746,7 +1730,7 @@ static void window_trans(AACContext * ac
int i;
if (ics->window_sequence != EIGHT_SHORT_SEQUENCE) {
- ff_imdct_calc(&ac->mdct, buf, in, out); // out can be abused for now as a temp buffer
+ ff_imdct_calc(&ac->mdct, buf, in, out); // Out can be abused, for now, as a temp buffer.
if (ics->window_sequence != LONG_STOP_SEQUENCE) {
ac->dsp.vector_fmul_add_add(out, buf, lwindow_prev, saved, ac->add_bias, 1024, 1);
} else {
@@ -1969,7 +1953,7 @@ static void transform_coupling_tool(AACC
cc_trans(ac, cc, &ac->che[ID_CPE][coup->tag_select[c]]->ch[1], index++);
} else {
av_log(ac->avccontext, AV_LOG_ERROR,
- "Coupling target %sE[%d] not available\n",
+ "coupling target %sE[%d] not available\n",
coup->is_cpe[c] ? "CP" : "SC", coup->tag_select[c]);
break;
}
@@ -2037,7 +2021,7 @@ static int output_samples(AVCodecContext
}
if(ac->mm[MIXDOWN_CENTER]) {
- /* Matrix mix-down */
+ /* matrix mix-down */
l = ac->mm[MIXDOWN_FRONT ]->ch[0].ret;
r = ac->mm[MIXDOWN_FRONT ]->ch[1].ret;
c = ac->mm[MIXDOWN_CENTER]->ch[0].ret;
Modified: aac/aactab.h
==============================================================================
--- aac/aactab.h (original)
+++ aac/aactab.h Tue Jul 1 12:12:10 2008
@@ -112,8 +112,8 @@ static const uint8_t num_swb_1024[] = {
};
static const uint16_t *swb_offset_128[] = {
- /* the last entry on the following row is swb_offset_128_64 but is a
- duplicate of swb_offset_128_96 */
+ /* The last entry on the following row is swb_offset_128_64 but is a
+ duplicate of swb_offset_128_96. */
swb_offset_128_96, swb_offset_128_96, swb_offset_128_96,
swb_offset_128_48, swb_offset_128_48, swb_offset_128_48,
swb_offset_128_24, swb_offset_128_24, swb_offset_128_16,
@@ -1072,3 +1072,4 @@ static const float pow2sf_tab[316] = {
#endif /* CONFIG_HARDCODED_TABLES */
#endif /* FFMPEG_AACTAB_H */
+
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