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usbaudio: send through dsp (new)
Does not seem to affect UI usability, but allowable DSP loads will vary based on device and playback sample rate. To-Do (someday): - Add dedicated DSP channel - How to apply DSP settings to above new channel? (UI) - How to lock out timestretch from being enabled? Change-Id: Ia24d1055340354e2c32e6008e7e2b03a8e88867d
This commit is contained in:
parent
7c4293af64
commit
c533222851
12 changed files with 94 additions and 31 deletions
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@ -249,7 +249,7 @@ static void codec_pcmbuf_insert_callback(
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}
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else
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{
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dsp_process(ci.dsp, &src, &dst);
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dsp_process(ci.dsp, &src, &dst, true);
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if (dst.remcount > 0)
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{
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@ -751,7 +751,7 @@ struct plugin_api {
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unsigned int setting, intptr_t value);
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struct dsp_config * (*dsp_get_config)(unsigned int dsp_id);
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void (*dsp_process)(struct dsp_config *dsp, struct dsp_buffer *src,
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struct dsp_buffer *dst);
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struct dsp_buffer *dst, bool thread_yield);
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enum channel_status (*mixer_channel_status)(enum pcm_mixer_channel channel);
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const void * (*mixer_channel_get_buffer)(enum pcm_mixer_channel channel,
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@ -444,7 +444,7 @@ static inline void synthbuf(void)
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dst.remcount = 0;
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dst.bufcount = available;
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dst.p16out = (int16_t *)outptr;
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rb->dsp_process(dsp, &src, &dst);
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rb->dsp_process(dsp, &src, &dst, true);
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if (dst.remcount > 0)
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{
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outptr += dst.remcount;
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@ -665,7 +665,7 @@ static void audio_thread(void)
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}
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dst.bufcount = size / (2 * sizeof (int16_t));
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rb->dsp_process(td.dsp, &td.src, &dst);
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rb->dsp_process(td.dsp, &td.src, &dst, true);
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if (dst.remcount > 0)
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{
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@ -230,7 +230,7 @@ static int process_dsp(const void *ch1, const void *ch2, int count)
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while (1)
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{
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int old_remcount = dst.remcount;
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rb->dsp_process(ci.dsp, &src, &dst);
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rb->dsp_process(ci.dsp, &src, &dst, true);
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if (dst.bufcount <= 0 ||
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(src.remcount <= 0 && dst.remcount <= old_remcount))
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@ -33,11 +33,14 @@ struct dsp_loop_context
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#endif
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};
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static inline void dsp_process_start(struct dsp_loop_context *ctx)
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static inline void dsp_process_start(struct dsp_loop_context *ctx, bool thread_yield)
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{
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/* At least perform one yield before starting */
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ctx->last_yield = current_tick;
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yield();
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if (thread_yield)
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{
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yield();
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}
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#if defined(CPU_COLDFIRE)
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/* set emac unit for dsp processing, and save old macsr, we're running in
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codec thread context at this point, so can't clobber it */
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@ -46,14 +49,17 @@ static inline void dsp_process_start(struct dsp_loop_context *ctx)
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#endif
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}
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static inline void dsp_process_loop(struct dsp_loop_context *ctx)
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static inline void dsp_process_loop(struct dsp_loop_context *ctx, bool thread_yield)
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{
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/* Yield at least once each tick */
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long tick = current_tick;
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if (TIME_AFTER(tick, ctx->last_yield))
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{
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ctx->last_yield = tick;
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yield();
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if (thread_yield)
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{
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yield();
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}
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}
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}
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@ -66,12 +72,12 @@ static inline void dsp_process_end(struct dsp_loop_context *ctx)
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(void)ctx;
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}
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#define DSP_PROCESS_START() \
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#define DSP_PROCESS_START(yield) \
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struct dsp_loop_context __ctx; \
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dsp_process_start(&__ctx)
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dsp_process_start(&__ctx, yield)
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#define DSP_PROCESS_LOOP() \
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dsp_process_loop(&__ctx)
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#define DSP_PROCESS_LOOP(yield) \
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dsp_process_loop(&__ctx, yield)
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#define DSP_PROCESS_END() \
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dsp_process_end(&__ctx)
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@ -511,7 +511,7 @@ static enum voice_state voice_buffer_insert(struct voice_thread_data *td)
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dst.bufcount = VOICE_PCM_FRAME_COUNT;
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td->dst = &dst;
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dsp_process(td->dsp, &td->src, &dst);
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dsp_process(td->dsp, &td->src, &dst, true);
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td->dst = NULL;
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voice_buf_commit(dst.remcount);
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@ -664,11 +664,12 @@ int32_t dsp_get_timestretch(void)
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\return
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\description
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void dsp_process(struct dsp_config *dsp, struct dsp_buffer *src, struct dsp_buffer *dst)
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void dsp_process(struct dsp_config *dsp, struct dsp_buffer *src, struct dsp_buffer *dst, bool thread_yield)
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\group sound
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\param dsp
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\param src
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\param dst
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\param thread_yield
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\description
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void dsp_set_crossfeed_type(int type)
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@ -42,6 +42,7 @@
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#include "settings.h"
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#include "core_alloc.h"
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#include "pcm_mixer.h"
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#include "dsp_core.h"
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#define LOGF_ENABLE
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#include "logf.h"
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@ -352,7 +353,7 @@ int tmp_saved_vol;
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static unsigned char *rx_buffer;
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int rx_buffer_handle;
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/* buffer size */
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static int rx_buf_size[NR_BUFFERS];
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static int rx_buf_size[NR_BUFFERS]; // only used for debug screen counter now
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/* index of the next buffer to play */
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static int rx_play_idx;
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/* index of the next buffer to fill */
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@ -362,6 +363,14 @@ bool playback_audio_underflow;
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/* usb overflow ? */
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bool usb_rx_overflow;
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/* dsp processing buffers */
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#define DSP_BUF_SIZE (BUFFER_SIZE*4) // arbitrarily x4
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#define REAL_DSP_BUF_SIZE ALIGN_UP(DSP_BUF_SIZE, 32)
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static uint16_t *dsp_buf;
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int dsp_buf_handle;
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static int dsp_buf_size[NR_BUFFERS];
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struct dsp_config *dsp = NULL;
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/* feedback variables */
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#define USB_FRAME_MAX 0x7FF
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#define NR_SAMPLES_HISTORY 32
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@ -502,7 +511,7 @@ int usb_audio_request_buf(void)
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audio_stop();
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// attempt to allocate the receive buffers
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rx_buffer_handle = core_alloc(NR_BUFFERS * REAL_BUF_SIZE);
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rx_buffer_handle = core_alloc(REAL_BUF_SIZE);
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if (rx_buffer_handle < 0)
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{
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alloc_failed = true;
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@ -518,6 +527,23 @@ int usb_audio_request_buf(void)
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// get the pointer to the actual buffer location
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rx_buffer = core_get_data(rx_buffer_handle);
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}
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dsp_buf_handle = core_alloc(NR_BUFFERS * REAL_DSP_BUF_SIZE);
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if (dsp_buf_handle < 0)
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{
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alloc_failed = true;
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rx_buffer_handle = core_free(rx_buffer_handle);
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rx_buffer = NULL;
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return -1;
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}
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else
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{
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alloc_failed = false;
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core_pin(dsp_buf_handle);
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dsp_buf = core_get_data(dsp_buf_handle);
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}
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// logf("usbaudio: got buffer");
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return 0;
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}
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@ -527,6 +553,9 @@ void usb_audio_free_buf(void)
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// logf("usbaudio: free buffer");
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rx_buffer_handle = core_free(rx_buffer_handle);
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rx_buffer = NULL;
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dsp_buf_handle = core_free(dsp_buf_handle);
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dsp_buf = NULL;
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}
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int usb_audio_request_endpoints(struct usb_class_driver *drv)
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@ -644,10 +673,11 @@ static void playback_audio_get_more(const void **start, size_t *size)
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*size = 0;
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return;
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}
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/* give buffer and advance */
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logf("usbaudio: buf adv");
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*start = rx_buffer + (rx_play_idx * REAL_BUF_SIZE);
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*size = rx_buf_size[rx_play_idx];
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*start = dsp_buf + (rx_play_idx * REAL_DSP_BUF_SIZE/sizeof(*dsp_buf));
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*size = dsp_buf_size[rx_play_idx];
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rx_play_idx = (rx_play_idx + 1) % NR_BUFFERS;
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/* if usb RX buffers had overflowed, we can start to receive again
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@ -658,7 +688,7 @@ static void playback_audio_get_more(const void **start, size_t *size)
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{
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logf("usbaudio: recover usb rx overflow");
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usb_rx_overflow = false;
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usb_drv_recv_nonblocking(out_iso_ep_adr, rx_buffer + (rx_usb_idx * REAL_BUF_SIZE), BUFFER_SIZE);
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usb_drv_recv_nonblocking(out_iso_ep_adr, rx_buffer, BUFFER_SIZE);
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}
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restore_irq(oldlevel);
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}
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@ -697,7 +727,8 @@ static void usb_audio_start_playback(void)
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mixer_set_frequency(hw_freq_sampr[as_playback_freq_idx]);
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pcm_apply_settings();
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mixer_channel_set_amplitude(PCM_MIXER_CHAN_USBAUDIO, MIX_AMP_UNITY);
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usb_drv_recv_nonblocking(out_iso_ep_adr, rx_buffer + (rx_usb_idx * REAL_BUF_SIZE), BUFFER_SIZE);
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usb_drv_recv_nonblocking(out_iso_ep_adr, rx_buffer, BUFFER_SIZE);
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}
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static void usb_audio_stop_playback(void)
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@ -1150,6 +1181,15 @@ void usb_audio_init_connection(void)
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{
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logf("usbaudio: init connection");
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dsp = dsp_get_config(CODEC_IDX_AUDIO);
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dsp_configure(dsp, DSP_RESET, 0);
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dsp_configure(dsp, DSP_SET_STEREO_MODE, STEREO_INTERLEAVED);
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dsp_configure(dsp, DSP_SET_SAMPLE_DEPTH, 16);
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#ifdef HAVE_PITCHCONTROL
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sound_set_pitch(PITCH_SPEED_100);
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dsp_set_timestretch(PITCH_SPEED_100);
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#endif
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usb_as_playback_intf_alt = 0;
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set_playback_sampling_frequency(HW_SAMPR_DEFAULT);
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tmp_saved_vol = sound_current(SOUND_VOLUME);
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@ -1261,20 +1301,36 @@ bool usb_audio_fast_transfer_complete(int ep, int dir, int status, int length)
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logf("usbaudio: frame: %d bytes: %d", usb_drv_get_frame_number(), length);
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if(status != 0)
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return true; /* FIXME how to handle error here ? */
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/* store length, queue buffer */
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rx_buf_size[rx_usb_idx] = length;
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rx_usb_idx = (rx_usb_idx + 1) % NR_BUFFERS;
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// debug screen counter
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samples_received = samples_received + length;
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// process through DSP right away!
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struct dsp_buffer src;
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src.remcount = length/4; // in samples
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src.pin[0] = rx_buffer;
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src.proc_mask = 0;
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struct dsp_buffer dst;
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dst.remcount = 0;
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dst.bufcount = DSP_BUF_SIZE/4; // in samples
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dst.p16out = dsp_buf + (rx_usb_idx * REAL_DSP_BUF_SIZE/sizeof(*dsp_buf)); // array index
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dsp_process(dsp, &src, &dst, false);
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dsp_buf_size[rx_usb_idx] = dst.remcount * 2 * sizeof(*dsp_buf); // need value in bytes
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rx_usb_idx = (rx_usb_idx + 1) % NR_BUFFERS;
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/* guard against IRQ to avoid race with completion audio completion */
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int oldlevel = disable_irq_save();
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/* setup a new transaction except if we ran out of buffers */
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if(rx_usb_idx != rx_play_idx)
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{
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logf("usbaudio: new transaction");
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usb_drv_recv_nonblocking(out_iso_ep_adr, rx_buffer + (rx_usb_idx*REAL_BUF_SIZE), BUFFER_SIZE);
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usb_drv_recv_nonblocking(out_iso_ep_adr, rx_buffer, BUFFER_SIZE);
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}
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else
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{
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@ -37,8 +37,8 @@
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#ifndef DSP_PROCESS_START
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/* These do nothing if not previously defined */
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#define DSP_PROCESS_START()
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#define DSP_PROCESS_LOOP()
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#define DSP_PROCESS_START(yield)
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#define DSP_PROCESS_LOOP(yield)
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#define DSP_PROCESS_END()
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#endif /* !DSP_PROCESS_START */
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@ -433,7 +433,7 @@ static FORCE_INLINE void dsp_proc_call(struct dsp_proc_slot *s,
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* of the call and how they function internally.
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*/
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void dsp_process(struct dsp_config *dsp, struct dsp_buffer *src,
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struct dsp_buffer *dst)
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struct dsp_buffer *dst, bool thread_yield)
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{
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if (dst->bufcount <= 0)
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{
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@ -441,7 +441,7 @@ void dsp_process(struct dsp_config *dsp, struct dsp_buffer *src,
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return;
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}
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DSP_PROCESS_START();
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DSP_PROCESS_START(thread_yield);
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/* Tag input with codec-specified sample format */
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src->format = dsp->io_data.format;
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@ -479,7 +479,7 @@ void dsp_process(struct dsp_config *dsp, struct dsp_buffer *src,
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dsp_advance_buffer32(buf, outcount);
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dsp_advance_buffer_output(dst, outcount);
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DSP_PROCESS_LOOP();
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DSP_PROCESS_LOOP(thread_yield);
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} /* while */
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DSP_PROCESS_END();
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@ -136,7 +136,7 @@ unsigned int dsp_get_id(const struct dsp_config *dsp);
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/* Process the given buffer - see implementation in dsp.c for more */
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void dsp_process(struct dsp_config *dsp, struct dsp_buffer *src,
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struct dsp_buffer *dst);
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struct dsp_buffer *dst, bool thread_yield);
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/* Change DSP settings */
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intptr_t dsp_configure(struct dsp_config *dsp, unsigned int setting,
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@ -455,7 +455,7 @@ static void ci_pcmbuf_insert(const void *ch1, const void *ch2, int count)
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dst.p16out = buf;
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dst.bufcount = out_count;
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dsp_process(ci.dsp, &src, &dst);
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dsp_process(ci.dsp, &src, &dst, true);
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if (dst.remcount > 0) {
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if (mode == MODE_WRITE)
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