forked from len0rd/rockbox
Make speex the new voice format for SWCODEC targets (non-Archos). Remove codec swapping and build speex voice decoding directly into the core binary.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@15668 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Michael Sevakis
parent
75432619e8
commit
99617d71ba
29 changed files with 753 additions and 1153 deletions
290
apps/dsp.c
290
apps/dsp.c
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@ -162,6 +162,10 @@ struct dsp_config
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int sample_bytes;
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int stereo_mode;
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int frac_bits;
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#ifdef HAVE_SW_TONE_CONTROLS
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/* Filter struct for software bass/treble controls */
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struct eqfilter tone_filter;
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#endif
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/* Functions that change depending upon settings - NULL if stage is
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disabled */
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sample_input_fn_type input_samples;
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@ -171,6 +175,7 @@ struct dsp_config
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way */
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channels_process_dsp_fn_type apply_gain;
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channels_process_fn_type apply_crossfeed;
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channels_process_fn_type eq_process;
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channels_process_fn_type channels_process;
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};
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@ -187,13 +192,13 @@ struct crossfeed_data crossfeed_data IDATA_ATTR = /* A */
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};
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/* Equalizer */
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static struct eq_state eq_data; /* A/V */
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static struct eq_state eq_data; /* A */
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/* Software tone controls */
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#ifdef HAVE_SW_TONE_CONTROLS
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static int prescale;
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static int bass;
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static int treble;
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/* Filter struct for software bass/treble controls */
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static struct eqfilter tone_filter;
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static int prescale; /* A/V */
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static int bass; /* A/V */
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static int treble; /* A/V */
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#endif
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/* Settings applicable to audio codec only */
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@ -202,7 +207,6 @@ static int channels_mode;
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long dsp_sw_gain;
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long dsp_sw_cross;
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static bool dither_enabled;
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static bool eq_enabled IBSS_ATTR;
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static long eq_precut;
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static long track_gain;
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static bool new_gain;
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@ -212,9 +216,8 @@ static long album_peak;
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static long replaygain;
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static bool crossfeed_enabled;
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#define audio_dsp (&dsp_conf[CODEC_IDX_AUDIO])
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#define voice_dsp (&dsp_conf[CODEC_IDX_VOICE])
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static struct dsp_config *dsp IDATA_ATTR = audio_dsp;
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#define audio_dsp (dsp_conf[CODEC_IDX_AUDIO])
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#define voice_dsp (dsp_conf[CODEC_IDX_VOICE])
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/* The internal format is 32-bit samples, non-interleaved, stereo. This
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* format is similar to the raw output from several codecs, so the amount
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@ -224,14 +227,6 @@ static struct dsp_config *dsp IDATA_ATTR = audio_dsp;
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int32_t sample_buf[SAMPLE_BUF_COUNT] IBSS_ATTR;
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static int32_t resample_buf[RESAMPLE_BUF_COUNT] IBSS_ATTR;
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/* set a new dsp and return old one */
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static inline struct dsp_config * switch_dsp(struct dsp_config *_dsp)
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{
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struct dsp_config * old_dsp = dsp;
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dsp = _dsp;
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return old_dsp;
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}
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#if 0
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/* Clip sample to arbitrary limits where range > 0 and min + range = max */
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static inline long clip_sample(int32_t sample, int32_t min, int32_t range)
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@ -263,8 +258,8 @@ int sound_get_pitch(void)
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void sound_set_pitch(int permille)
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{
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pitch_ratio = permille;
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dsp_configure(DSP_SWITCH_FREQUENCY, dsp->codec_frequency);
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dsp_configure(&audio_dsp, DSP_SWITCH_FREQUENCY,
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audio_dsp.codec_frequency);
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}
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/* Convert count samples to the internal format, if needed. Updates src
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@ -386,7 +381,7 @@ static void sample_input_gt_native_ni_stereo(
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* * dsp->stereo_mode (A/V)
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* * dsp->sample_depth (A/V)
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*/
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static void sample_input_new_format(void)
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static void sample_input_new_format(struct dsp_config *dsp)
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{
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static const sample_input_fn_type sample_input_functions[] =
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{
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@ -462,7 +457,7 @@ static void sample_output_dithered(int count, struct dsp_data *data,
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int ch;
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int16_t *d;
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for (ch = 0; ch < dsp->data.num_channels; ch++)
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for (ch = 0; ch < data->num_channels; ch++)
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{
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struct dither_data * const dither = &dither_data[ch];
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int32_t *s = src[ch];
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@ -505,7 +500,7 @@ static void sample_output_dithered(int count, struct dsp_data *data,
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}
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}
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if (dsp->data.num_channels == 2)
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if (data->num_channels == 2)
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return;
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/* Have to duplicate left samples into the right channel since
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@ -530,7 +525,7 @@ static void sample_output_dithered(int count, struct dsp_data *data,
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* * dsp->stereo_mode (A/V)
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* * dither_enabled (A)
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*/
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static void sample_output_new_format(void)
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static void sample_output_new_format(struct dsp_config *dsp)
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{
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static const sample_output_fn_type sample_output_functions[] =
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{
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@ -542,7 +537,7 @@ static void sample_output_new_format(void)
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int out = dsp->data.num_channels - 1;
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if (dsp == audio_dsp && dither_enabled)
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if (dsp == &audio_dsp && dither_enabled)
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out += 2;
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dsp->output_samples = sample_output_functions[out];
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@ -638,7 +633,7 @@ static int dsp_upsample(int count, struct dsp_data *data,
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}
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#endif /* DSP_HAVE_ASM_RESAMPLING */
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static void resampler_new_delta(void)
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static void resampler_new_delta(struct dsp_config *dsp)
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{
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dsp->data.resample_data.delta = (unsigned long)
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dsp->frequency * 65536LL / NATIVE_FREQUENCY;
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@ -663,7 +658,7 @@ static void resampler_new_delta(void)
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* done, to refer to the resampled data. Returns number of stereo samples
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* for further processing.
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*/
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static inline int resample(int count, int32_t *src[])
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static inline int resample(struct dsp_config *dsp, int count, int32_t *src[])
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{
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int32_t *dst[2] =
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{
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@ -679,12 +674,8 @@ static inline int resample(int count, int32_t *src[])
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return count;
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}
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static void dither_init(void)
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static void dither_init(struct dsp_config *dsp)
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{
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/* Voice codec should not reset the audio codec's dither data */
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if (dsp != audio_dsp)
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return;
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memset(dither_data, 0, sizeof (dither_data));
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dither_bias = (1L << (dsp->frac_bits - NATIVE_DEPTH));
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dither_mask = (1L << (dsp->frac_bits + 1 - NATIVE_DEPTH)) - 1;
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@ -692,11 +683,9 @@ static void dither_init(void)
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void dsp_dither_enable(bool enable)
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{
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/* Be sure audio dsp is current to set correct function */
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struct dsp_config *old_dsp = switch_dsp(audio_dsp);
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struct dsp_config *dsp = &audio_dsp;
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dither_enabled = enable;
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sample_output_new_format();
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switch_dsp(old_dsp);
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sample_output_new_format(dsp);
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}
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/* Applies crossfeed to the stereo signal in src.
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@ -762,9 +751,8 @@ static void apply_crossfeed(int count, int32_t *buf[])
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void dsp_set_crossfeed(bool enable)
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{
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crossfeed_enabled = enable;
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audio_dsp->apply_crossfeed =
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(enable && audio_dsp->data.num_channels > 1)
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? apply_crossfeed : NULL;
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audio_dsp.apply_crossfeed = (enable && audio_dsp.data.num_channels > 1)
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? apply_crossfeed : NULL;
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}
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void dsp_set_crossfeed_direct_gain(int gain)
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@ -830,12 +818,12 @@ static void set_gain(struct dsp_config *dsp)
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dsp->data.gain = DEFAULT_GAIN;
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/* Replay gain not relevant to voice */
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if (dsp == audio_dsp && replaygain)
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if (dsp == &audio_dsp && replaygain)
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{
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dsp->data.gain = replaygain;
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}
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if (eq_enabled && eq_precut)
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if (dsp->eq_process && eq_precut)
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{
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dsp->data.gain =
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(long) (((int64_t) dsp->data.gain * eq_precut) >> 24);
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@ -853,16 +841,6 @@ static void set_gain(struct dsp_config *dsp)
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dsp->apply_gain = dsp->data.gain != 0 ? dsp_apply_gain : NULL;
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}
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/**
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* Use to enable the equalizer.
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*
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* @param enable true to enable the equalizer
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*/
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void dsp_set_eq(bool enable)
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{
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eq_enabled = enable;
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}
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/**
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* Update the amount to cut the audio before applying the equalizer.
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*
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@ -871,8 +849,7 @@ void dsp_set_eq(bool enable)
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void dsp_set_eq_precut(int precut)
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{
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eq_precut = get_replaygain_int(precut * -10);
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set_gain(audio_dsp);
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set_gain(voice_dsp); /* For EQ precut */
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set_gain(&audio_dsp);
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}
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/**
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@ -929,7 +906,7 @@ static void eq_process(int count, int32_t *buf[])
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EQ_PEAK_SHIFT, /* peaking */
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EQ_SHELF_SHIFT, /* high shelf */
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};
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unsigned int channels = dsp->data.num_channels;
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unsigned int channels = audio_dsp.data.num_channels;
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int i;
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/* filter configuration currently is 1 low shelf filter, 3 band peaking
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@ -944,6 +921,17 @@ static void eq_process(int count, int32_t *buf[])
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}
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}
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/**
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* Use to enable the equalizer.
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*
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* @param enable true to enable the equalizer
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*/
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void dsp_set_eq(bool enable)
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{
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audio_dsp.eq_process = enable ? eq_process : NULL;
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set_gain(&audio_dsp);
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}
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void dsp_set_stereo_width(int value)
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{
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long width, straight, cross;
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@ -966,50 +954,6 @@ void dsp_set_stereo_width(int value)
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dsp_sw_cross = cross << 8;
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}
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#if CONFIG_CODEC == SWCODEC
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#ifdef HAVE_SW_TONE_CONTROLS
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static void set_tone_controls(void)
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{
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filter_bishelf_coefs(0xffffffff/NATIVE_FREQUENCY*200,
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0xffffffff/NATIVE_FREQUENCY*3500,
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bass, treble, -prescale, tone_filter.coefs);
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}
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#endif
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/* Hook back from firmware/ part of audio, which can't/shouldn't call apps/
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* code directly.
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*/
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int dsp_callback(int msg, intptr_t param)
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{
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switch (msg) {
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#ifdef HAVE_SW_TONE_CONTROLS
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case DSP_CALLBACK_SET_PRESCALE:
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prescale = param;
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set_tone_controls();
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break;
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/* prescaler is always set after calling any of these, so we wait with
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* calculating coefs until the above case is hit.
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*/
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case DSP_CALLBACK_SET_BASS:
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bass = param;
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break;
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case DSP_CALLBACK_SET_TREBLE:
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treble = param;
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#endif
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case DSP_CALLBACK_SET_CHANNEL_CONFIG:
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dsp_set_channel_config(param);
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break;
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case DSP_CALLBACK_SET_STEREO_WIDTH:
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dsp_set_stereo_width(param);
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break;
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default:
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break;
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}
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return 0;
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}
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#endif
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/**
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* Implements the different channel configurations and stereo width.
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*/
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@ -1098,14 +1042,64 @@ void dsp_set_channel_config(int value)
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};
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if ((unsigned)value >= ARRAYLEN(channels_process_functions) ||
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audio_dsp->stereo_mode == STEREO_MONO)
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audio_dsp.stereo_mode == STEREO_MONO)
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{
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value = SOUND_CHAN_STEREO;
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}
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/* This doesn't apply to voice */
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channels_mode = value;
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audio_dsp->channels_process = channels_process_functions[value];
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audio_dsp.channels_process = channels_process_functions[value];
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}
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#if CONFIG_CODEC == SWCODEC
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#ifdef HAVE_SW_TONE_CONTROLS
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static void set_tone_controls(void)
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{
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filter_bishelf_coefs(0xffffffff/NATIVE_FREQUENCY*200,
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0xffffffff/NATIVE_FREQUENCY*3500,
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bass, treble, -prescale,
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audio_dsp.tone_filter.coefs);
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/* Sync the voice dsp coefficients */
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memcpy(&voice_dsp.tone_filter.coefs, audio_dsp.tone_filter.coefs,
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sizeof (voice_dsp.tone_filter.coefs));
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}
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#endif
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/* Hook back from firmware/ part of audio, which can't/shouldn't call apps/
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* code directly.
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*/
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int dsp_callback(int msg, intptr_t param)
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{
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switch (msg) {
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#ifdef HAVE_SW_TONE_CONTROLS
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case DSP_CALLBACK_SET_PRESCALE:
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prescale = param;
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set_tone_controls();
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break;
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/* prescaler is always set after calling any of these, so we wait with
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* calculating coefs until the above case is hit.
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*/
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case DSP_CALLBACK_SET_BASS:
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bass = param;
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break;
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case DSP_CALLBACK_SET_TREBLE:
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treble = param;
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#endif
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case DSP_CALLBACK_SET_CHANNEL_CONFIG:
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dsp_set_channel_config(param);
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break;
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case DSP_CALLBACK_SET_STEREO_WIDTH:
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dsp_set_stereo_width(param);
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break;
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default:
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break;
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}
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return 0;
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}
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#endif
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/* Process and convert src audio to dst based on the DSP configuration,
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* reading count number of audio samples. dst is assumed to be large
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* enough; use dsp_output_count() to get the required number. src is an
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@ -1114,7 +1108,7 @@ void dsp_set_channel_config(int value)
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* non-interleaved stereo, it contains two pointers, one for each audio
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* channel. Returns number of bytes written to dst.
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*/
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int dsp_process(char *dst, const char *src[], int count)
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int dsp_process(struct dsp_config *dsp, char *dst, const char *src[], int count)
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{
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int32_t *tmp[2];
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int written = 0;
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@ -1142,25 +1136,19 @@ int dsp_process(char *dst, const char *src[], int count)
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if (dsp->apply_gain)
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dsp->apply_gain(samples, &dsp->data, tmp);
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if (dsp->resample && (samples = resample(samples, tmp)) <= 0)
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if (dsp->resample && (samples = resample(dsp, samples, tmp)) <= 0)
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break; /* I'm pretty sure we're downsampling here */
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if (dsp->apply_crossfeed)
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dsp->apply_crossfeed(samples, tmp);
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/* TODO: EQ and tone controls need separate structs for audio and voice
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* DSP processing thanks to filter history. isn't really audible now, but
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* might be the day we start handling voice more delicately. Planned
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* changes may well run all relevent channels through the same EQ so
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* perhaps not.
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*/
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if (eq_enabled)
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eq_process(samples, tmp);
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if (dsp->eq_process)
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dsp->eq_process(samples, tmp);
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#ifdef HAVE_SW_TONE_CONTROLS
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if ((bass | treble) != 0)
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eq_filter(tmp, &tone_filter, samples, dsp->data.num_channels,
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FILTER_BISHELF_SHIFT);
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eq_filter(tmp, &dsp->tone_filter, samples,
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dsp->data.num_channels, FILTER_BISHELF_SHIFT);
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#endif
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if (dsp->channels_process)
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@ -1187,7 +1175,7 @@ int dsp_process(char *dst, const char *src[], int count)
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* of the resampler).
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*/
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/* dsp_input_size MUST be called afterwards */
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int dsp_output_count(int count)
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int dsp_output_count(struct dsp_config *dsp, int count)
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{
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if (dsp->resample)
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{
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@ -1209,7 +1197,7 @@ int dsp_output_count(int count)
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/* Given count output samples, calculate number of input samples
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* that would be consumed in order to fill the output buffer.
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*/
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int dsp_input_count(int count)
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int dsp_input_count(struct dsp_config *dsp, int count)
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{
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/* count is now the number of resampled input samples. Convert to
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original input samples. */
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@ -1225,41 +1213,37 @@ int dsp_input_count(int count)
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return count;
|
||||
}
|
||||
|
||||
int dsp_stereo_mode(void)
|
||||
{
|
||||
return dsp->stereo_mode;
|
||||
}
|
||||
|
||||
static void dsp_set_gain_var(long *var, long value)
|
||||
{
|
||||
/* Voice shouldn't mess with these */
|
||||
if (dsp == audio_dsp)
|
||||
{
|
||||
*var = value;
|
||||
new_gain = true;
|
||||
}
|
||||
*var = value;
|
||||
new_gain = true;
|
||||
}
|
||||
|
||||
static void dsp_update_functions(void)
|
||||
static void dsp_update_functions(struct dsp_config *dsp)
|
||||
{
|
||||
sample_input_new_format();
|
||||
sample_output_new_format();
|
||||
if (dsp == audio_dsp)
|
||||
sample_input_new_format(dsp);
|
||||
sample_output_new_format(dsp);
|
||||
if (dsp == &audio_dsp)
|
||||
dsp_set_crossfeed(crossfeed_enabled);
|
||||
}
|
||||
|
||||
bool dsp_configure(int setting, intptr_t value)
|
||||
intptr_t dsp_configure(struct dsp_config *dsp, int setting, intptr_t value)
|
||||
{
|
||||
switch (setting)
|
||||
{
|
||||
case DSP_SWITCH_CODEC:
|
||||
if ((uintptr_t)value <= 1)
|
||||
switch_dsp(&dsp_conf[value]);
|
||||
break;
|
||||
case DSP_MYDSP:
|
||||
switch (value)
|
||||
{
|
||||
case CODEC_IDX_AUDIO:
|
||||
return (intptr_t)&audio_dsp;
|
||||
case CODEC_IDX_VOICE:
|
||||
return (intptr_t)&voice_dsp;
|
||||
default:
|
||||
return (intptr_t)NULL;
|
||||
}
|
||||
|
||||
case DSP_SET_FREQUENCY:
|
||||
memset(&dsp->data.resample_data, 0,
|
||||
sizeof (dsp->data.resample_data));
|
||||
memset(&dsp->data.resample_data, 0, sizeof (dsp->data.resample_data));
|
||||
/* Fall through!!! */
|
||||
case DSP_SWITCH_FREQUENCY:
|
||||
dsp->codec_frequency = (value == 0) ? NATIVE_FREQUENCY : value;
|
||||
|
|
@ -1267,12 +1251,12 @@ bool dsp_configure(int setting, intptr_t value)
|
|||
if we're called from the main audio thread. Voice UI thread should
|
||||
not need this feature.
|
||||
*/
|
||||
if (dsp == audio_dsp)
|
||||
if (dsp == &audio_dsp)
|
||||
dsp->frequency = pitch_ratio * dsp->codec_frequency / 1000;
|
||||
else
|
||||
dsp->frequency = dsp->codec_frequency;
|
||||
|
||||
resampler_new_delta();
|
||||
resampler_new_delta(dsp);
|
||||
break;
|
||||
|
||||
case DSP_SET_SAMPLE_DEPTH:
|
||||
|
|
@ -1294,14 +1278,14 @@ bool dsp_configure(int setting, intptr_t value)
|
|||
}
|
||||
|
||||
dsp->data.output_scale = dsp->frac_bits + 1 - NATIVE_DEPTH;
|
||||
sample_input_new_format();
|
||||
dither_init();
|
||||
sample_input_new_format(dsp);
|
||||
dither_init(dsp);
|
||||
break;
|
||||
|
||||
case DSP_SET_STEREO_MODE:
|
||||
dsp->stereo_mode = value;
|
||||
dsp->data.num_channels = value == STEREO_MONO ? 1 : 2;
|
||||
dsp_update_functions();
|
||||
dsp_update_functions(dsp);
|
||||
break;
|
||||
|
||||
case DSP_RESET:
|
||||
|
|
@ -1315,7 +1299,7 @@ bool dsp_configure(int setting, intptr_t value)
|
|||
dsp->data.clip_min = -((1 << WORD_FRACBITS));
|
||||
dsp->codec_frequency = dsp->frequency = NATIVE_FREQUENCY;
|
||||
|
||||
if (dsp == audio_dsp)
|
||||
if (dsp == &audio_dsp)
|
||||
{
|
||||
track_gain = 0;
|
||||
album_gain = 0;
|
||||
|
|
@ -1324,31 +1308,35 @@ bool dsp_configure(int setting, intptr_t value)
|
|||
new_gain = true;
|
||||
}
|
||||
|
||||
dsp_update_functions();
|
||||
resampler_new_delta();
|
||||
dsp_update_functions(dsp);
|
||||
resampler_new_delta(dsp);
|
||||
break;
|
||||
|
||||
case DSP_FLUSH:
|
||||
memset(&dsp->data.resample_data, 0,
|
||||
sizeof (dsp->data.resample_data));
|
||||
resampler_new_delta();
|
||||
dither_init();
|
||||
resampler_new_delta(dsp);
|
||||
dither_init(dsp);
|
||||
break;
|
||||
|
||||
case DSP_SET_TRACK_GAIN:
|
||||
dsp_set_gain_var(&track_gain, value);
|
||||
if (dsp == &audio_dsp)
|
||||
dsp_set_gain_var(&track_gain, value);
|
||||
break;
|
||||
|
||||
case DSP_SET_ALBUM_GAIN:
|
||||
dsp_set_gain_var(&album_gain, value);
|
||||
if (dsp == &audio_dsp)
|
||||
dsp_set_gain_var(&album_gain, value);
|
||||
break;
|
||||
|
||||
case DSP_SET_TRACK_PEAK:
|
||||
dsp_set_gain_var(&track_peak, value);
|
||||
if (dsp == &audio_dsp)
|
||||
dsp_set_gain_var(&track_peak, value);
|
||||
break;
|
||||
|
||||
case DSP_SET_ALBUM_PEAK:
|
||||
dsp_set_gain_var(&album_peak, value);
|
||||
if (dsp == &audio_dsp)
|
||||
dsp_set_gain_var(&album_peak, value);
|
||||
break;
|
||||
|
||||
default:
|
||||
|
|
@ -1404,5 +1392,5 @@ void dsp_set_replaygain(void)
|
|||
|
||||
/* Store in S8.23 format to simplify calculations. */
|
||||
replaygain = gain;
|
||||
set_gain(audio_dsp);
|
||||
set_gain(&audio_dsp);
|
||||
}
|
||||
|
|
|
|||
Loading…
Add table
Add a link
Reference in a new issue