diff options
Diffstat (limited to 'gnu/packages/patches/mswebrtc-b64-refactor.patch')
| -rw-r--r-- | gnu/packages/patches/mswebrtc-b64-refactor.patch | 949 |
1 files changed, 949 insertions, 0 deletions
diff --git a/gnu/packages/patches/mswebrtc-b64-refactor.patch b/gnu/packages/patches/mswebrtc-b64-refactor.patch new file mode 100644 index 0000000000..af4a1e1682 --- /dev/null +++ b/gnu/packages/patches/mswebrtc-b64-refactor.patch @@ -0,0 +1,949 @@ +From 17e72f00831a36da387ceafe7f3076ffa8f66aba Mon Sep 17 00:00:00 2001 +From: Clemence Him <clemence.him@belledonne-communications.com> +Date: Fri, 22 Sep 2023 14:28:02 +0200 +Subject: [PATCH] Base64 functions refactoring + +--- + aec.c | 781 +++++++++++++++++++++++++++++----------------------------- + 1 file changed, 394 insertions(+), 387 deletions(-) + +diff --git a/aec.c b/aec.c +index 271f370..995f655 100644 +--- a/aec.c ++++ b/aec.c +@@ -24,19 +24,18 @@ Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA. + #include "mediastreamer2/msfilter.h" + #include "mediastreamer2/msticker.h" + #ifdef BUILD_AEC +-#include "echo_cancellation.h" + #include "aec_splitting_filter.h" ++#include "echo_cancellation.h" + #endif + #ifdef BUILD_AECM + #include "echo_control_mobile.h" + #endif +-#include "ortp/b64.h" + + #ifdef _WIN32 + #include <malloc.h> /* for alloca */ + #endif + +-//#define EC_DUMP 1 ++// #define EC_DUMP 1 + #ifdef ANDROID + #define EC_DUMP_PREFIX "/sdcard" + #else +@@ -48,466 +47,485 @@ Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA. + static const float smooth_factor = 0.05f; + static const int framesize = 80; + +- + typedef enum _WebRTCAECType { +- WebRTCAECTypeNormal, +- WebRTCAECTypeMobile ++ WebRTCAECTypeNormal, ++ WebRTCAECTypeMobile + } WebRTCAECType; + + typedef struct WebRTCAECState { +- void *aecInst; +- MSBufferizer delayed_ref; +- MSFlowControlledBufferizer ref; +- MSBufferizer echo; +- int framesize; +- int samplerate; +- int delay_ms; +- int nominal_ref_samples; +- char *state_str; ++ void *aecInst; ++ MSBufferizer delayed_ref; ++ MSFlowControlledBufferizer ref; ++ MSBufferizer echo; ++ int framesize; ++ int samplerate; ++ int delay_ms; ++ int nominal_ref_samples; ++ char *state_str; + #ifdef EC_DUMP +- FILE *echofile; +- FILE *reffile; +- FILE *cleanfile; ++ FILE *echofile; ++ FILE *reffile; ++ FILE *cleanfile; + #endif +- bool_t echostarted; +- bool_t bypass_mode; +- bool_t using_zeroes; +- WebRTCAECType aec_type; ++ bool_t echostarted; ++ bool_t bypass_mode; ++ bool_t using_zeroes; ++ WebRTCAECType aec_type; + #ifdef BUILD_AEC +- MSWebRtcAecSplittingFilter *splitting_filter; ++ MSWebRtcAecSplittingFilter *splitting_filter; + #endif + } WebRTCAECState; + + static void webrtc_aecgeneric_init(MSFilter *f, WebRTCAECType aec_type) { +- WebRTCAECState *s = (WebRTCAECState *) ms_new0(WebRTCAECState, 1); +- +- s->samplerate = 8000; +- ms_bufferizer_init(&s->delayed_ref); +- ms_bufferizer_init(&s->echo); +- ms_flow_controlled_bufferizer_init(&s->ref, f, s->samplerate, 1); +- s->delay_ms = 0; +- s->aecInst = NULL; +- s->framesize = framesize; +- s->state_str = NULL; +- s->using_zeroes = FALSE; +- s->echostarted = FALSE; +- s->bypass_mode = FALSE; +- s->aec_type = aec_type; ++ WebRTCAECState *s = (WebRTCAECState *)ms_new0(WebRTCAECState, 1); ++ ++ s->samplerate = 8000; ++ ms_bufferizer_init(&s->delayed_ref); ++ ms_bufferizer_init(&s->echo); ++ ms_flow_controlled_bufferizer_init(&s->ref, f, s->samplerate, 1); ++ s->delay_ms = 0; ++ s->aecInst = NULL; ++ s->framesize = framesize; ++ s->state_str = NULL; ++ s->using_zeroes = FALSE; ++ s->echostarted = FALSE; ++ s->bypass_mode = FALSE; ++ s->aec_type = aec_type; + + #ifdef EC_DUMP +- { +- char *fname = ms_strdup_printf("%s/mswebrtcaec-%p-echo.raw", EC_DUMP_PREFIX, f); +- s->echofile = fopen(fname, "w"); +- ms_free(fname); +- fname = ms_strdup_printf("%s/mswebrtcaec-%p-ref.raw", EC_DUMP_PREFIX, f); +- s->reffile = fopen(fname, "w"); +- ms_free(fname); +- fname = ms_strdup_printf("%s/mswebrtcaec-%p-clean.raw", EC_DUMP_PREFIX, f); +- s->cleanfile = fopen(fname, "w"); +- ms_free(fname); +- } ++ { ++ char *fname = ++ ms_strdup_printf("%s/mswebrtcaec-%p-echo.raw", EC_DUMP_PREFIX, f); ++ s->echofile = fopen(fname, "w"); ++ ms_free(fname); ++ fname = ms_strdup_printf("%s/mswebrtcaec-%p-ref.raw", EC_DUMP_PREFIX, f); ++ s->reffile = fopen(fname, "w"); ++ ms_free(fname); ++ fname = ms_strdup_printf("%s/mswebrtcaec-%p-clean.raw", EC_DUMP_PREFIX, f); ++ s->cleanfile = fopen(fname, "w"); ++ ms_free(fname); ++ } + #endif + +- f->data = s; ++ f->data = s; + } + + #ifdef BUILD_AEC + static void webrtc_aec_init(MSFilter *f) { +- webrtc_aecgeneric_init(f, WebRTCAECTypeNormal); ++ webrtc_aecgeneric_init(f, WebRTCAECTypeNormal); + } + #endif + + #ifdef BUILD_AECM + static void webrtc_aecm_init(MSFilter *f) { +- webrtc_aecgeneric_init(f, WebRTCAECTypeMobile); ++ webrtc_aecgeneric_init(f, WebRTCAECTypeMobile); + } + #endif + + static void webrtc_aec_uninit(MSFilter *f) { +- WebRTCAECState *s = (WebRTCAECState *) f->data; +- if (s->state_str) ms_free(s->state_str); +- ms_bufferizer_uninit(&s->delayed_ref); ++ WebRTCAECState *s = (WebRTCAECState *)f->data; ++ if (s->state_str) ++ ms_free(s->state_str); ++ ms_bufferizer_uninit(&s->delayed_ref); + #ifdef EC_DUMP +- if (s->echofile) +- fclose(s->echofile); +- if (s->reffile) +- fclose(s->reffile); ++ if (s->echofile) ++ fclose(s->echofile); ++ if (s->reffile) ++ fclose(s->reffile); + #endif +- ms_free(s); ++ ms_free(s); + } + + static void configure_flow_controlled_bufferizer(WebRTCAECState *s) { +- ms_flow_controlled_bufferizer_set_samplerate(&s->ref, s->samplerate); +- ms_flow_controlled_bufferizer_set_max_size_ms(&s->ref, s->delay_ms); +- ms_flow_controlled_bufferizer_set_granularity_ms(&s->ref, (s->framesize * 1000) / s->samplerate); ++ ms_flow_controlled_bufferizer_set_samplerate(&s->ref, s->samplerate); ++ ms_flow_controlled_bufferizer_set_max_size_ms(&s->ref, s->delay_ms); ++ ms_flow_controlled_bufferizer_set_granularity_ms( ++ &s->ref, (s->framesize * 1000) / s->samplerate); + } + + static void webrtc_aec_preprocess(MSFilter *f) { +- WebRTCAECState *s = (WebRTCAECState *) f->data; ++ WebRTCAECState *s = (WebRTCAECState *)f->data; + #ifdef BUILD_AEC +- AecConfig aec_config; ++ AecConfig aec_config; + #endif + #ifdef BUILD_AECM +- AecmConfig aecm_config; +- int error_code; ++ AecmConfig aecm_config; ++ int error_code; + #endif +- int delay_samples = 0; +- mblk_t *m; ++ int delay_samples = 0; ++ mblk_t *m; + +- s->echostarted = FALSE; +- delay_samples = s->delay_ms * s->samplerate / 1000; +- s->framesize=(framesize*s->samplerate)/8000; +- ms_message("Initializing WebRTC echo canceler with framesize=%i, delay_ms=%i, delay_samples=%i", s->framesize, s->delay_ms, delay_samples); +- configure_flow_controlled_bufferizer(s); ++ s->echostarted = FALSE; ++ delay_samples = s->delay_ms * s->samplerate / 1000; ++ s->framesize = (framesize * s->samplerate) / 8000; ++ ms_message("Initializing WebRTC echo canceler with framesize=%i, " ++ "delay_ms=%i, delay_samples=%i", ++ s->framesize, s->delay_ms, delay_samples); ++ configure_flow_controlled_bufferizer(s); + + #ifdef BUILD_AEC +- if (s->aec_type == WebRTCAECTypeNormal) { +- if ((s->aecInst = WebRtcAec_Create()) == NULL) { +- s->bypass_mode = TRUE; +- ms_error("WebRtcAec_Create(): error, entering bypass mode"); +- return; +- } +- if ((WebRtcAec_Init(s->aecInst, MIN(48000, s->samplerate), s->samplerate)) < 0) { +- ms_error("WebRtcAec_Init(): WebRTC echo canceller does not support %d samplerate", s->samplerate); +- s->bypass_mode = TRUE; +- ms_error("Entering bypass mode"); +- return; +- } +- aec_config.nlpMode = kAecNlpAggressive; +- aec_config.skewMode = kAecFalse; +- aec_config.metricsMode = kAecFalse; +- aec_config.delay_logging = kAecFalse; +- if (WebRtcAec_set_config(s->aecInst, aec_config) != 0) { +- ms_error("WebRtcAec_set_config(): failed."); +- } +- } ++ if (s->aec_type == WebRTCAECTypeNormal) { ++ if ((s->aecInst = WebRtcAec_Create()) == NULL) { ++ s->bypass_mode = TRUE; ++ ms_error("WebRtcAec_Create(): error, entering bypass mode"); ++ return; ++ } ++ if ((WebRtcAec_Init(s->aecInst, MIN(48000, s->samplerate), s->samplerate)) < ++ 0) { ++ ms_error("WebRtcAec_Init(): WebRTC echo canceller does not support %d " ++ "samplerate", ++ s->samplerate); ++ s->bypass_mode = TRUE; ++ ms_error("Entering bypass mode"); ++ return; ++ } ++ aec_config.nlpMode = kAecNlpAggressive; ++ aec_config.skewMode = kAecFalse; ++ aec_config.metricsMode = kAecFalse; ++ aec_config.delay_logging = kAecFalse; ++ if (WebRtcAec_set_config(s->aecInst, aec_config) != 0) { ++ ms_error("WebRtcAec_set_config(): failed."); ++ } ++ } + #endif + #ifdef BUILD_AECM +- if (s->aec_type == WebRTCAECTypeMobile) { +- if ((s->aecInst = WebRtcAecm_Create()) == NULL) { +- s->bypass_mode = TRUE; +- ms_error("WebRtcAecm_Create(): error, entering bypass mode"); +- return; +- } +- if ((error_code = WebRtcAecm_Init(s->aecInst, s->samplerate)) < 0) { +- if (error_code == AECM_BAD_PARAMETER_ERROR) { +- ms_error("WebRtcAecm_Init(): WebRTC echo canceller does not support %d samplerate", s->samplerate); +- } +- s->bypass_mode = TRUE; +- ms_error("Entering bypass mode"); +- return; +- } +- aecm_config.cngMode = TRUE; +- aecm_config.echoMode = 3; +- if (WebRtcAecm_set_config(s->aecInst, aecm_config)!=0){ +- ms_error("WebRtcAecm_set_config(): failed."); +- } +- } ++ if (s->aec_type == WebRTCAECTypeMobile) { ++ if ((s->aecInst = WebRtcAecm_Create()) == NULL) { ++ s->bypass_mode = TRUE; ++ ms_error("WebRtcAecm_Create(): error, entering bypass mode"); ++ return; ++ } ++ if ((error_code = WebRtcAecm_Init(s->aecInst, s->samplerate)) < 0) { ++ if (error_code == AECM_BAD_PARAMETER_ERROR) { ++ ms_error("WebRtcAecm_Init(): WebRTC echo canceller does not support %d " ++ "samplerate", ++ s->samplerate); ++ } ++ s->bypass_mode = TRUE; ++ ms_error("Entering bypass mode"); ++ return; ++ } ++ aecm_config.cngMode = TRUE; ++ aecm_config.echoMode = 3; ++ if (WebRtcAecm_set_config(s->aecInst, aecm_config) != 0) { ++ ms_error("WebRtcAecm_set_config(): failed."); ++ } ++ } + #endif + +- /* fill with zeroes for the time of the delay*/ +- m = allocb(delay_samples * 2, 0); +- m->b_wptr += delay_samples * 2; +- ms_bufferizer_put(&s->delayed_ref, m); +- s->nominal_ref_samples = delay_samples; ++ /* fill with zeroes for the time of the delay*/ ++ m = allocb(delay_samples * 2, 0); ++ m->b_wptr += delay_samples * 2; ++ ms_bufferizer_put(&s->delayed_ref, m); ++ s->nominal_ref_samples = delay_samples; + } + + /* inputs[0]= reference signal from far end (sent to soundcard) + * inputs[1]= near speech & echo signal (read from soundcard) + * outputs[0]= is a copy of inputs[0] to be sent to soundcard + * outputs[1]= near end speech, echo removed - towards far end +-*/ ++ */ + static void webrtc_aec_process(MSFilter *f) { +- WebRTCAECState *s = (WebRTCAECState *) f->data; +- int nbytes = s->framesize * sizeof(int16_t); +- mblk_t *refm; +- int16_t *ref, *echo; +- int nbands = 1; +- int bandsize = s->framesize; +- +- if (s->bypass_mode) { +- while ((refm = ms_queue_get(f->inputs[0])) != NULL) { +- ms_queue_put(f->outputs[0], refm); +- } +- while ((refm = ms_queue_get(f->inputs[1])) != NULL) { +- ms_queue_put(f->outputs[1], refm); +- } +- return; +- } +- +- if (f->inputs[0] != NULL) { +- if (s->echostarted) { +- while ((refm = ms_queue_get(f->inputs[0])) != NULL) { +- mblk_t *cp=dupmsg(refm); +- ms_bufferizer_put(&s->delayed_ref,cp); +- ms_flow_controlled_bufferizer_put(&s->ref,refm); +- } +- } else { +- ms_warning("Getting reference signal but no echo to synchronize on."); +- ms_queue_flush(f->inputs[0]); +- } +- } +- +- ms_bufferizer_put_from_queue(&s->echo, f->inputs[1]); +- +- ref = (int16_t *) alloca(nbytes); +- echo = (int16_t *) alloca(nbytes); ++ WebRTCAECState *s = (WebRTCAECState *)f->data; ++ int nbytes = s->framesize * sizeof(int16_t); ++ mblk_t *refm; ++ int16_t *ref, *echo; ++ int nbands = 1; ++ int bandsize = s->framesize; ++ ++ if (s->bypass_mode) { ++ while ((refm = ms_queue_get(f->inputs[0])) != NULL) { ++ ms_queue_put(f->outputs[0], refm); ++ } ++ while ((refm = ms_queue_get(f->inputs[1])) != NULL) { ++ ms_queue_put(f->outputs[1], refm); ++ } ++ return; ++ } ++ ++ if (f->inputs[0] != NULL) { ++ if (s->echostarted) { ++ while ((refm = ms_queue_get(f->inputs[0])) != NULL) { ++ mblk_t *cp = dupmsg(refm); ++ ms_bufferizer_put(&s->delayed_ref, cp); ++ ms_flow_controlled_bufferizer_put(&s->ref, refm); ++ } ++ } else { ++ ms_warning("Getting reference signal but no echo to synchronize on."); ++ ms_queue_flush(f->inputs[0]); ++ } ++ } ++ ++ ms_bufferizer_put_from_queue(&s->echo, f->inputs[1]); ++ ++ ref = (int16_t *)alloca(nbytes); ++ echo = (int16_t *)alloca(nbytes); + #ifdef BUILD_AEC +- if (s->aec_type == WebRTCAECTypeNormal) { +- if (s->samplerate > 16000) { +- nbands = s->samplerate / 16000; +- bandsize = 160; +- } +- if (!s->splitting_filter) { +- s->splitting_filter = mswebrtc_aec_splitting_filter_create(nbands, bandsize); +- } +- } ++ if (s->aec_type == WebRTCAECTypeNormal) { ++ if (s->samplerate > 16000) { ++ nbands = s->samplerate / 16000; ++ bandsize = 160; ++ } ++ if (!s->splitting_filter) { ++ s->splitting_filter = ++ mswebrtc_aec_splitting_filter_create(nbands, bandsize); ++ } ++ } + #endif +- while (ms_bufferizer_read(&s->echo, (uint8_t *)echo, (size_t)nbytes) >= (size_t)nbytes) { +- mblk_t *oecho = allocb(nbytes, 0); +- int avail; +- int avail_samples; +- +- if (!s->echostarted) s->echostarted = TRUE; +- if ((avail = ms_bufferizer_get_avail(&s->delayed_ref)) < ((s->nominal_ref_samples * 2) + nbytes)) { +- /*we don't have enough to read in a reference signal buffer, inject silence instead*/ +- refm = allocb(nbytes, 0); +- memset(refm->b_wptr, 0, nbytes); +- refm->b_wptr += nbytes; +- ms_bufferizer_put(&s->delayed_ref, refm); +- /* +- * However, we don't inject this silence buffer to the sound card, in order to break the following bad loop: +- * - the sound playback filter detects it has too many pending samples, then triggers an event to request samples to be dropped upstream. +- * - the upstream MSFlowControl filter is requested to drop samples, which it starts to do. +- * - necessarily shortly after the AEC goes into a situation where it has not enough reference samples while processing an audio buffer from mic. +- * - if the AEC injects a silence buffer as output, then it will RECREATE a situation where the sound playback filter has too many pending samples. +- * That's why we should not do this. +- * By not doing this, we will create a discrepancy between what we really injected to the soundcard, and what we told to the +- * echo canceller about the samples we injected. This shifts the echo. The echo canceller will re-converge quickly to take into +- * account the situation. +- * +- */ +- //ms_queue_put(f->outputs[0], dupmsg(refm)); +- if (!s->using_zeroes) { +- ms_warning("Not enough ref samples, using zeroes"); +- s->using_zeroes = TRUE; +- } +- } else { +- if (s->using_zeroes) { +- ms_message("Samples are back."); +- s->using_zeroes = FALSE; +- } +- /* read from our no-delay buffer and output */ +- refm = allocb(nbytes, 0); +- if (ms_flow_controlled_bufferizer_read(&s->ref, refm->b_wptr, nbytes) == 0) { +- ms_fatal("Should never happen"); +- } +- refm->b_wptr += nbytes; +- ms_queue_put(f->outputs[0], refm); +- } +- +- /*now read a valid buffer of delayed ref samples*/ +- if (ms_bufferizer_read(&s->delayed_ref, (uint8_t *)ref, nbytes) == 0) { +- ms_fatal("Should never happen"); +- } +- avail -= nbytes; +- avail_samples = avail / 2; ++ while (ms_bufferizer_read(&s->echo, (uint8_t *)echo, (size_t)nbytes) >= ++ (size_t)nbytes) { ++ mblk_t *oecho = allocb(nbytes, 0); ++ int avail; ++ int avail_samples; ++ ++ if (!s->echostarted) ++ s->echostarted = TRUE; ++ if ((avail = ms_bufferizer_get_avail(&s->delayed_ref)) < ++ ((s->nominal_ref_samples * 2) + nbytes)) { ++ /*we don't have enough to read in a reference signal buffer, inject ++ * silence instead*/ ++ refm = allocb(nbytes, 0); ++ memset(refm->b_wptr, 0, nbytes); ++ refm->b_wptr += nbytes; ++ ms_bufferizer_put(&s->delayed_ref, refm); ++ /* ++ * However, we don't inject this silence buffer to the sound card, in ++ * order to break the following bad loop: ++ * - the sound playback filter detects it has too many pending samples, ++ * then triggers an event to request samples to be dropped upstream. ++ * - the upstream MSFlowControl filter is requested to drop samples, which ++ * it starts to do. ++ * - necessarily shortly after the AEC goes into a situation where it has ++ * not enough reference samples while processing an audio buffer from mic. ++ * - if the AEC injects a silence buffer as output, then it will RECREATE ++ * a situation where the sound playback filter has too many pending ++ * samples. That's why we should not do this. By not doing this, we will ++ * create a discrepancy between what we really injected to the soundcard, ++ * and what we told to the echo canceller about the samples we injected. ++ * This shifts the echo. The echo canceller will re-converge quickly to ++ * take into account the situation. ++ * ++ */ ++ // ms_queue_put(f->outputs[0], dupmsg(refm)); ++ if (!s->using_zeroes) { ++ ms_warning("Not enough ref samples, using zeroes"); ++ s->using_zeroes = TRUE; ++ } ++ } else { ++ if (s->using_zeroes) { ++ ms_message("Samples are back."); ++ s->using_zeroes = FALSE; ++ } ++ /* read from our no-delay buffer and output */ ++ refm = allocb(nbytes, 0); ++ if (ms_flow_controlled_bufferizer_read(&s->ref, refm->b_wptr, nbytes) == ++ 0) { ++ ms_fatal("Should never happen"); ++ } ++ refm->b_wptr += nbytes; ++ ms_queue_put(f->outputs[0], refm); ++ } ++ ++ /*now read a valid buffer of delayed ref samples*/ ++ if (ms_bufferizer_read(&s->delayed_ref, (uint8_t *)ref, nbytes) == 0) { ++ ms_fatal("Should never happen"); ++ } ++ avail -= nbytes; ++ avail_samples = avail / 2; + + #ifdef EC_DUMP +- if (s->reffile) +- fwrite(ref, nbytes, 1, s->reffile); +- if (s->echofile) +- fwrite(echo, nbytes, 1, s->echofile); ++ if (s->reffile) ++ fwrite(ref, nbytes, 1, s->reffile); ++ if (s->echofile) ++ fwrite(echo, nbytes, 1, s->echofile); + #endif + #ifdef BUILD_AEC +- if (s->aec_type == WebRTCAECTypeNormal) { +- mswebrtc_aec_splitting_filter_analysis(s->splitting_filter, ref, echo); +- if (WebRtcAec_BufferFarend(s->aecInst, +- mswebrtc_aec_splitting_filter_get_ref(s->splitting_filter), +- (size_t)mswebrtc_aec_splitting_filter_get_bandsize(s->splitting_filter)) != 0) +- ms_error("WebRtcAec_BufferFarend() failed."); +- if (WebRtcAec_Process(s->aecInst, +- mswebrtc_aec_splitting_filter_get_echo_bands(s->splitting_filter), +- mswebrtc_aec_splitting_filter_get_number_of_bands(s->splitting_filter), +- mswebrtc_aec_splitting_filter_get_output_bands(s->splitting_filter), +- (size_t)mswebrtc_aec_splitting_filter_get_bandsize(s->splitting_filter), 0, 0) != 0) +- ms_error("WebRtcAec_Process() failed."); +- mswebrtc_aec_splitting_filter_synthesis(s->splitting_filter, (int16_t *)oecho->b_wptr); +- } ++ if (s->aec_type == WebRTCAECTypeNormal) { ++ mswebrtc_aec_splitting_filter_analysis(s->splitting_filter, ref, echo); ++ if (WebRtcAec_BufferFarend( ++ s->aecInst, ++ mswebrtc_aec_splitting_filter_get_ref(s->splitting_filter), ++ (size_t)mswebrtc_aec_splitting_filter_get_bandsize( ++ s->splitting_filter)) != 0) ++ ms_error("WebRtcAec_BufferFarend() failed."); ++ if (WebRtcAec_Process( ++ s->aecInst, ++ mswebrtc_aec_splitting_filter_get_echo_bands(s->splitting_filter), ++ mswebrtc_aec_splitting_filter_get_number_of_bands( ++ s->splitting_filter), ++ mswebrtc_aec_splitting_filter_get_output_bands( ++ s->splitting_filter), ++ (size_t)mswebrtc_aec_splitting_filter_get_bandsize( ++ s->splitting_filter), ++ 0, 0) != 0) ++ ms_error("WebRtcAec_Process() failed."); ++ mswebrtc_aec_splitting_filter_synthesis(s->splitting_filter, ++ (int16_t *)oecho->b_wptr); ++ } + #endif + #ifdef BUILD_AECM +- if (s->aec_type == WebRTCAECTypeMobile) { +- if (WebRtcAecm_BufferFarend(s->aecInst, ref, (size_t)s->framesize) != 0) +- ms_error("WebRtcAecm_BufferFarend() failed."); +- if (WebRtcAecm_Process(s->aecInst, echo, NULL, (int16_t *)oecho->b_wptr, (size_t)s->framesize, 0) != 0) +- ms_error("WebRtcAecm_Process() failed."); +- } ++ if (s->aec_type == WebRTCAECTypeMobile) { ++ if (WebRtcAecm_BufferFarend(s->aecInst, ref, (size_t)s->framesize) != 0) ++ ms_error("WebRtcAecm_BufferFarend() failed."); ++ if (WebRtcAecm_Process(s->aecInst, echo, NULL, (int16_t *)oecho->b_wptr, ++ (size_t)s->framesize, 0) != 0) ++ ms_error("WebRtcAecm_Process() failed."); ++ } + #endif + #ifdef EC_DUMP +- if (s->cleanfile) +- fwrite(oecho->b_wptr, nbytes, 1, s->cleanfile); ++ if (s->cleanfile) ++ fwrite(oecho->b_wptr, nbytes, 1, s->cleanfile); + #endif +- oecho->b_wptr += nbytes; +- ms_queue_put(f->outputs[1], oecho); +- } ++ oecho->b_wptr += nbytes; ++ ms_queue_put(f->outputs[1], oecho); ++ } + } + + static void webrtc_aec_postprocess(MSFilter *f) { +- WebRTCAECState *s = (WebRTCAECState *) f->data; ++ WebRTCAECState *s = (WebRTCAECState *)f->data; + +- ms_bufferizer_flush(&s->delayed_ref); +- ms_bufferizer_flush(&s->echo); +- ms_flow_controlled_bufferizer_flush(&s->ref); ++ ms_bufferizer_flush(&s->delayed_ref); ++ ms_bufferizer_flush(&s->echo); ++ ms_flow_controlled_bufferizer_flush(&s->ref); + #ifdef BUILD_AEC +- if (s->splitting_filter) { +- mswebrtc_aec_splitting_filter_destroy(s->splitting_filter); +- s->splitting_filter = NULL; +- } ++ if (s->splitting_filter) { ++ mswebrtc_aec_splitting_filter_destroy(s->splitting_filter); ++ s->splitting_filter = NULL; ++ } + #endif +- if (s->aecInst != NULL) { ++ if (s->aecInst != NULL) { + #ifdef BUILD_AEC +- if (s->aec_type == WebRTCAECTypeNormal) { +- WebRtcAec_Free(s->aecInst); +- } ++ if (s->aec_type == WebRTCAECTypeNormal) { ++ WebRtcAec_Free(s->aecInst); ++ } + #endif + #ifdef BUILD_AECM +- if (s->aec_type == WebRTCAECTypeMobile) { +- WebRtcAecm_Free(s->aecInst); +- } ++ if (s->aec_type == WebRTCAECTypeMobile) { ++ WebRtcAecm_Free(s->aecInst); ++ } + #endif +- s->aecInst = NULL; +- } ++ s->aecInst = NULL; ++ } + } + + static int webrtc_aec_set_sr(MSFilter *f, void *arg) { +- WebRTCAECState *s = (WebRTCAECState *) f->data; +- int requested_sr = *(int *) arg; +- int sr = requested_sr; +- +- if ((s->aec_type == WebRTCAECTypeNormal) && (requested_sr >= 48000)) { +- sr = 48000; +- } else if ((s->aec_type == WebRTCAECTypeNormal) && (requested_sr >= 32000)) { +- sr = 32000; +- } else if (requested_sr >= 16000) { +- sr = 16000; +- } else { +- sr = 8000; +- } +- if (sr != requested_sr) +- ms_message("Webrtc %s does not support sampling rate %i, using %i instead", ((s->aec_type == WebRTCAECTypeNormal)?"aec":"aecm"),requested_sr, sr); +- +- s->samplerate = sr; +- configure_flow_controlled_bufferizer(s); +- return 0; ++ WebRTCAECState *s = (WebRTCAECState *)f->data; ++ int requested_sr = *(int *)arg; ++ int sr = requested_sr; ++ ++ if ((s->aec_type == WebRTCAECTypeNormal) && (requested_sr >= 48000)) { ++ sr = 48000; ++ } else if ((s->aec_type == WebRTCAECTypeNormal) && (requested_sr >= 32000)) { ++ sr = 32000; ++ } else if (requested_sr >= 16000) { ++ sr = 16000; ++ } else { ++ sr = 8000; ++ } ++ if (sr != requested_sr) ++ ms_message("Webrtc %s does not support sampling rate %i, using %i instead", ++ ((s->aec_type == WebRTCAECTypeNormal) ? "aec" : "aecm"), ++ requested_sr, sr); ++ ++ s->samplerate = sr; ++ configure_flow_controlled_bufferizer(s); ++ return 0; + } + + static int webrtc_aec_get_sr(MSFilter *f, void *arg) { +- WebRTCAECState *s = (WebRTCAECState *) f->data; +- *(int *) arg=s->samplerate; +- return 0; ++ WebRTCAECState *s = (WebRTCAECState *)f->data; ++ *(int *)arg = s->samplerate; ++ return 0; + } + + static int webrtc_aec_set_framesize(MSFilter *f, void *arg) { +- /* Do nothing because the WebRTC echo canceller only accept specific values: 80 and 160. We use 80 at 8khz, and 160 at 16khz */ +- return 0; ++ /* Do nothing because the WebRTC echo canceller only accept specific values: ++ * 80 and 160. We use 80 at 8khz, and 160 at 16khz */ ++ return 0; + } + + static int webrtc_aec_set_delay(MSFilter *f, void *arg) { +- WebRTCAECState *s = (WebRTCAECState *) f->data; +- s->delay_ms = *(int *) arg; +- configure_flow_controlled_bufferizer(s); +- return 0; ++ WebRTCAECState *s = (WebRTCAECState *)f->data; ++ s->delay_ms = *(int *)arg; ++ configure_flow_controlled_bufferizer(s); ++ return 0; + } + + static int webrtc_aec_set_tail_length(MSFilter *f, void *arg) { +- /* Do nothing because this is not needed by the WebRTC echo canceller. */ +- return 0; ++ /* Do nothing because this is not needed by the WebRTC echo canceller. */ ++ return 0; + } + static int webrtc_aec_set_bypass_mode(MSFilter *f, void *arg) { +- WebRTCAECState *s = (WebRTCAECState *) f->data; +- s->bypass_mode = *(bool_t *) arg; +- ms_message("set EC bypass mode to [%i]", s->bypass_mode); +- return 0; ++ WebRTCAECState *s = (WebRTCAECState *)f->data; ++ s->bypass_mode = *(bool_t *)arg; ++ ms_message("set EC bypass mode to [%i]", s->bypass_mode); ++ return 0; + } + static int webrtc_aec_get_bypass_mode(MSFilter *f, void *arg) { +- WebRTCAECState *s = (WebRTCAECState *) f->data; +- *(bool_t *) arg = s->bypass_mode; +- return 0; ++ WebRTCAECState *s = (WebRTCAECState *)f->data; ++ *(bool_t *)arg = s->bypass_mode; ++ return 0; + } + + static int webrtc_aec_set_state(MSFilter *f, void *arg) { +- WebRTCAECState *s = (WebRTCAECState *) f->data; +- s->state_str = ms_strdup((const char *) arg); +- return 0; ++ WebRTCAECState *s = (WebRTCAECState *)f->data; ++ s->state_str = ms_strdup((const char *)arg); ++ return 0; + } + + static int webrtc_aec_get_state(MSFilter *f, void *arg) { +- WebRTCAECState *s = (WebRTCAECState *) f->data; +- *(char **) arg = s->state_str; +- return 0; ++ WebRTCAECState *s = (WebRTCAECState *)f->data; ++ *(char **)arg = s->state_str; ++ return 0; + } + + static MSFilterMethod webrtc_aec_methods[] = { +- { MS_FILTER_SET_SAMPLE_RATE , webrtc_aec_set_sr }, +- { MS_FILTER_GET_SAMPLE_RATE , webrtc_aec_get_sr }, +- { MS_ECHO_CANCELLER_SET_TAIL_LENGTH , webrtc_aec_set_tail_length }, +- { MS_ECHO_CANCELLER_SET_DELAY , webrtc_aec_set_delay }, +- { MS_ECHO_CANCELLER_SET_FRAMESIZE , webrtc_aec_set_framesize }, +- { MS_ECHO_CANCELLER_SET_BYPASS_MODE , webrtc_aec_set_bypass_mode }, +- { MS_ECHO_CANCELLER_GET_BYPASS_MODE , webrtc_aec_get_bypass_mode }, +- { MS_ECHO_CANCELLER_GET_STATE_STRING , webrtc_aec_get_state }, +- { MS_ECHO_CANCELLER_SET_STATE_STRING , webrtc_aec_set_state }, +- { 0, NULL } +-}; +- ++ {MS_FILTER_SET_SAMPLE_RATE, webrtc_aec_set_sr}, ++ {MS_FILTER_GET_SAMPLE_RATE, webrtc_aec_get_sr}, ++ {MS_ECHO_CANCELLER_SET_TAIL_LENGTH, webrtc_aec_set_tail_length}, ++ {MS_ECHO_CANCELLER_SET_DELAY, webrtc_aec_set_delay}, ++ {MS_ECHO_CANCELLER_SET_FRAMESIZE, webrtc_aec_set_framesize}, ++ {MS_ECHO_CANCELLER_SET_BYPASS_MODE, webrtc_aec_set_bypass_mode}, ++ {MS_ECHO_CANCELLER_GET_BYPASS_MODE, webrtc_aec_get_bypass_mode}, ++ {MS_ECHO_CANCELLER_GET_STATE_STRING, webrtc_aec_get_state}, ++ {MS_ECHO_CANCELLER_SET_STATE_STRING, webrtc_aec_set_state}, ++ {0, NULL}}; + + #ifdef BUILD_AEC + +-#define MS_WEBRTC_AEC_NAME "MSWebRTCAEC" ++#define MS_WEBRTC_AEC_NAME "MSWebRTCAEC" + #define MS_WEBRTC_AEC_DESCRIPTION "Echo canceller using WebRTC library." +-#define MS_WEBRTC_AEC_CATEGORY MS_FILTER_OTHER +-#define MS_WEBRTC_AEC_ENC_FMT NULL +-#define MS_WEBRTC_AEC_NINPUTS 2 +-#define MS_WEBRTC_AEC_NOUTPUTS 2 +-#define MS_WEBRTC_AEC_FLAGS 0 ++#define MS_WEBRTC_AEC_CATEGORY MS_FILTER_OTHER ++#define MS_WEBRTC_AEC_ENC_FMT NULL ++#define MS_WEBRTC_AEC_NINPUTS 2 ++#define MS_WEBRTC_AEC_NOUTPUTS 2 ++#define MS_WEBRTC_AEC_FLAGS 0 + + #ifdef _MSC_VER + + MSFilterDesc ms_webrtc_aec_desc = { +- MS_FILTER_PLUGIN_ID, +- MS_WEBRTC_AEC_NAME, +- MS_WEBRTC_AEC_DESCRIPTION, +- MS_WEBRTC_AEC_CATEGORY, +- MS_WEBRTC_AEC_ENC_FMT, +- MS_WEBRTC_AEC_NINPUTS, +- MS_WEBRTC_AEC_NOUTPUTS, +- webrtc_aec_init, +- webrtc_aec_preprocess, +- webrtc_aec_process, +- webrtc_aec_postprocess, +- webrtc_aec_uninit, +- webrtc_aec_methods, +- MS_WEBRTC_AEC_FLAGS +-}; ++ MS_FILTER_PLUGIN_ID, MS_WEBRTC_AEC_NAME, MS_WEBRTC_AEC_DESCRIPTION, ++ MS_WEBRTC_AEC_CATEGORY, MS_WEBRTC_AEC_ENC_FMT, MS_WEBRTC_AEC_NINPUTS, ++ MS_WEBRTC_AEC_NOUTPUTS, webrtc_aec_init, webrtc_aec_preprocess, ++ webrtc_aec_process, webrtc_aec_postprocess, webrtc_aec_uninit, ++ webrtc_aec_methods, MS_WEBRTC_AEC_FLAGS}; + + #else + +-MSFilterDesc ms_webrtc_aec_desc = { +- .id = MS_FILTER_PLUGIN_ID, +- .name = MS_WEBRTC_AEC_NAME, +- .text = MS_WEBRTC_AEC_DESCRIPTION, +- .category = MS_WEBRTC_AEC_CATEGORY, +- .enc_fmt = MS_WEBRTC_AEC_ENC_FMT, +- .ninputs = MS_WEBRTC_AEC_NINPUTS, +- .noutputs = MS_WEBRTC_AEC_NOUTPUTS, +- .init = webrtc_aec_init, +- .preprocess = webrtc_aec_preprocess, +- .process = webrtc_aec_process, +- .postprocess = webrtc_aec_postprocess, +- .uninit = webrtc_aec_uninit, +- .methods = webrtc_aec_methods, +- .flags = MS_WEBRTC_AEC_FLAGS +-}; ++MSFilterDesc ms_webrtc_aec_desc = {.id = MS_FILTER_PLUGIN_ID, ++ .name = MS_WEBRTC_AEC_NAME, ++ .text = MS_WEBRTC_AEC_DESCRIPTION, ++ .category = MS_WEBRTC_AEC_CATEGORY, ++ .enc_fmt = MS_WEBRTC_AEC_ENC_FMT, ++ .ninputs = MS_WEBRTC_AEC_NINPUTS, ++ .noutputs = MS_WEBRTC_AEC_NOUTPUTS, ++ .init = webrtc_aec_init, ++ .preprocess = webrtc_aec_preprocess, ++ .process = webrtc_aec_process, ++ .postprocess = webrtc_aec_postprocess, ++ .uninit = webrtc_aec_uninit, ++ .methods = webrtc_aec_methods, ++ .flags = MS_WEBRTC_AEC_FLAGS}; + + #endif + +@@ -517,51 +535,40 @@ MS_FILTER_DESC_EXPORT(ms_webrtc_aec_desc) + + #ifdef BUILD_AECM + +-#define MS_WEBRTC_AECM_NAME "MSWebRTCAECM" +-#define MS_WEBRTC_AECM_DESCRIPTION "Echo canceller for mobile using WebRTC library." +-#define MS_WEBRTC_AECM_CATEGORY MS_FILTER_OTHER +-#define MS_WEBRTC_AECM_ENC_FMT NULL +-#define MS_WEBRTC_AECM_NINPUTS 2 +-#define MS_WEBRTC_AECM_NOUTPUTS 2 +-#define MS_WEBRTC_AECM_FLAGS 0 ++#define MS_WEBRTC_AECM_NAME "MSWebRTCAECM" ++#define MS_WEBRTC_AECM_DESCRIPTION \ ++ "Echo canceller for mobile using WebRTC library." ++#define MS_WEBRTC_AECM_CATEGORY MS_FILTER_OTHER ++#define MS_WEBRTC_AECM_ENC_FMT NULL ++#define MS_WEBRTC_AECM_NINPUTS 2 ++#define MS_WEBRTC_AECM_NOUTPUTS 2 ++#define MS_WEBRTC_AECM_FLAGS 0 + + #ifdef _MSC_VER + + MSFilterDesc ms_webrtc_aecm_desc = { +- MS_FILTER_PLUGIN_ID, +- MS_WEBRTC_AECM_NAME, +- MS_WEBRTC_AECM_DESCRIPTION, +- MS_WEBRTC_AECM_CATEGORY, +- MS_WEBRTC_AECM_ENC_FMT, +- MS_WEBRTC_AECM_NINPUTS, +- MS_WEBRTC_AECM_NOUTPUTS, +- webrtc_aecm_init, +- webrtc_aec_preprocess, +- webrtc_aec_process, +- webrtc_aec_postprocess, +- webrtc_aec_uninit, +- webrtc_aec_methods, +- MS_WEBRTC_AECM_FLAGS +-}; ++ MS_FILTER_PLUGIN_ID, MS_WEBRTC_AECM_NAME, MS_WEBRTC_AECM_DESCRIPTION, ++ MS_WEBRTC_AECM_CATEGORY, MS_WEBRTC_AECM_ENC_FMT, MS_WEBRTC_AECM_NINPUTS, ++ MS_WEBRTC_AECM_NOUTPUTS, webrtc_aecm_init, webrtc_aec_preprocess, ++ webrtc_aec_process, webrtc_aec_postprocess, webrtc_aec_uninit, ++ webrtc_aec_methods, MS_WEBRTC_AECM_FLAGS}; + + #else + +-MSFilterDesc ms_webrtc_aecm_desc = { +- .id = MS_FILTER_PLUGIN_ID, +- .name = MS_WEBRTC_AECM_NAME, +- .text = MS_WEBRTC_AECM_DESCRIPTION, +- .category = MS_WEBRTC_AECM_CATEGORY, +- .enc_fmt = MS_WEBRTC_AECM_ENC_FMT, +- .ninputs = MS_WEBRTC_AECM_NINPUTS, +- .noutputs = MS_WEBRTC_AECM_NOUTPUTS, +- .init = webrtc_aecm_init, +- .preprocess = webrtc_aec_preprocess, +- .process = webrtc_aec_process, +- .postprocess = webrtc_aec_postprocess, +- .uninit = webrtc_aec_uninit, +- .methods = webrtc_aec_methods, +- .flags = MS_WEBRTC_AECM_FLAGS +-}; ++MSFilterDesc ms_webrtc_aecm_desc = {.id = MS_FILTER_PLUGIN_ID, ++ .name = MS_WEBRTC_AECM_NAME, ++ .text = MS_WEBRTC_AECM_DESCRIPTION, ++ .category = MS_WEBRTC_AECM_CATEGORY, ++ .enc_fmt = MS_WEBRTC_AECM_ENC_FMT, ++ .ninputs = MS_WEBRTC_AECM_NINPUTS, ++ .noutputs = MS_WEBRTC_AECM_NOUTPUTS, ++ .init = webrtc_aecm_init, ++ .preprocess = webrtc_aec_preprocess, ++ .process = webrtc_aec_process, ++ .postprocess = webrtc_aec_postprocess, ++ .uninit = webrtc_aec_uninit, ++ .methods = webrtc_aec_methods, ++ .flags = MS_WEBRTC_AECM_FLAGS}; + + #endif + +-- +GitLab + |