Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff

-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
   using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
   and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate.  This allows
   for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
   updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.

-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c

Review: https://reviewboard.asterisk.org/r/1104/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 64 insertions, 143 deletions

diff --git a/codecs/codec_resample.c b/codecs/codec_resample.c
index 80823bb7c..20bbc5c9b 100644
--- a/codecs/codec_resample.c
+++ b/codecs/codec_resample.c
@@ -1,9 +1,10 @@
 /*
  * Asterisk -- An open source telephony toolkit.
  *
- * Copyright (C) 2007, Digium, Inc.
+ * Copyright (C) 2011, Digium, Inc.
  *
  * Russell Bryant <russell@digium.com>
+ * David Vossel <dvossel@digium.com>
  *
  * See http://www.asterisk.org for more information about
  * the Asterisk project. Please do not directly contact
@@ -21,9 +22,6 @@
  *
  * \brief Resample slinear audio
  * 
- * \note To install libresample, check it out of the following repository:
- * <code>$ svn co http://svn.digium.com/svn/thirdparty/libresample/trunk</code>
- *
  * \ingroup codecs
  */
 
@@ -32,170 +30,75 @@
  ***/
 
 #include "asterisk.h"
+#include "speex/speex_resampler.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
-/* These are for SHRT_MAX and FLT_MAX -- { */
-#if defined(__Darwin__) || defined(__OpenBSD__) || defined(__FreeBSD__) || defined(__NetBSD__) || defined(__CYGWIN__)
-#include <float.h>
-#else
-#include <values.h>
-#endif
-#include <limits.h>
-/* } */
-
-#include <libresample.h>
-
 #include "asterisk/module.h"
 #include "asterisk/translate.h"
-
 #include "asterisk/slin.h"
 
-#define RESAMPLER_QUALITY 1
-
 #define OUTBUF_SIZE   8096
 
-struct slin16_to_slin8_pvt {
-	void *resampler;
-	float resample_factor;
-};
-
-struct slin8_to_slin16_pvt {
-	void *resampler;
-	float resample_factor;
+static struct ast_translator *translators;
+static int trans_size;
+static int id_list[] = {
+	AST_FORMAT_SLINEAR,
+	AST_FORMAT_SLINEAR12,
+	AST_FORMAT_SLINEAR16,
+	AST_FORMAT_SLINEAR24,
+	AST_FORMAT_SLINEAR32,
+	AST_FORMAT_SLINEAR44,
+	AST_FORMAT_SLINEAR48,
+	AST_FORMAT_SLINEAR96,
+	AST_FORMAT_SLINEAR192,
 };
 
-static int slin16_to_slin8_new(struct ast_trans_pvt *pvt)
+static int resamp_new(struct ast_trans_pvt *pvt)
 {
-	struct slin16_to_slin8_pvt *resamp_pvt = pvt->pvt;
+	int err;
 
-	resamp_pvt->resample_factor = 8000.0 / 16000.0;
-
-	if (!(resamp_pvt->resampler = resample_open(RESAMPLER_QUALITY, resamp_pvt->resample_factor, resamp_pvt->resample_factor)))
-		return -1;
-
-	return 0;
-}
-
-static int slin8_to_slin16_new(struct ast_trans_pvt *pvt)
-{
-	struct slin8_to_slin16_pvt *resamp_pvt = pvt->pvt;
-
-	resamp_pvt->resample_factor = 16000.0 / 8000.0;
-
-	if (!(resamp_pvt->resampler = resample_open(RESAMPLER_QUALITY, resamp_pvt->resample_factor, resamp_pvt->resample_factor)))
+	if (!(pvt->pvt = speex_resampler_init(1, ast_format_rate(&pvt->t->src_format), ast_format_rate(&pvt->t->dst_format), 5, &err))) {
 		return -1;
+	}
 
 	return 0;
 }
 
-static void slin16_to_slin8_destroy(struct ast_trans_pvt *pvt)
-{
-	struct slin16_to_slin8_pvt *resamp_pvt = pvt->pvt;
-
-	if (resamp_pvt->resampler)
-		resample_close(resamp_pvt->resampler);
-}
-
-static void slin8_to_slin16_destroy(struct ast_trans_pvt *pvt)
+static void resamp_destroy(struct ast_trans_pvt *pvt)
 {
-	struct slin8_to_slin16_pvt *resamp_pvt = pvt->pvt;
-
-	if (resamp_pvt->resampler)
-		resample_close(resamp_pvt->resampler);
+	SpeexResamplerState *resamp_pvt = pvt->pvt;
+	speex_resampler_destroy(resamp_pvt);
 }
 
-static int resample_frame(struct ast_trans_pvt *pvt,
-	void *resampler, float resample_factor, struct ast_frame *f)
+static int resamp_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
 {
-	int total_in_buf_used = 0;
-	int total_out_buf_used = 0;
-	int16_t *in_buf = (int16_t *) f->data.ptr;
-	int16_t *out_buf = pvt->outbuf.i16 + pvt->samples;
-	float in_buf_f[f->samples];
-	float out_buf_f[2048];
-	int res = 0;
-	int i;
-
-	for (i = 0; i < f->samples; i++)
-		in_buf_f[i] = in_buf[i] * (FLT_MAX / SHRT_MAX);
-
-	while (total_in_buf_used < f->samples) {
-		int in_buf_used, out_buf_used;
+	SpeexResamplerState *resamp_pvt = pvt->pvt;
+	unsigned int out_samples = (OUTBUF_SIZE / sizeof(int16_t)) - pvt->samples;
+	unsigned int in_samples = f->samples;
 
-		out_buf_used = resample_process(resampler, resample_factor,
-			&in_buf_f[total_in_buf_used], f->samples - total_in_buf_used,
-			0, &in_buf_used,
-			&out_buf_f[total_out_buf_used], ARRAY_LEN(out_buf_f) - total_out_buf_used);
+	speex_resampler_process_int(resamp_pvt,
+		0,
+		f->data.ptr,
+		&in_samples,
+		pvt->outbuf.i16 + pvt->samples,
+		&out_samples);
 
-		if (out_buf_used < 0)
-			break;
+	pvt->samples += out_samples;
+	pvt->datalen += out_samples * 2;
 
-		total_out_buf_used += out_buf_used;
-		total_in_buf_used += in_buf_used;
-
-		if (total_out_buf_used == ARRAY_LEN(out_buf_f)) {
-			ast_log(LOG_ERROR, "Output buffer filled ... need to increase its size\n");
-			res = -1;
-			break;
-		}
-	}
-
-	for (i = 0; i < total_out_buf_used; i++)
-		out_buf[i] = out_buf_f[i] * (SHRT_MAX / FLT_MAX);	
-
-	pvt->samples += total_out_buf_used;
-	pvt->datalen += (total_out_buf_used * sizeof(int16_t));
-
-	return res;
-}
-
-static int slin16_to_slin8_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
-{
-	struct slin16_to_slin8_pvt *resamp_pvt = pvt->pvt;
-	void *resampler = resamp_pvt->resampler;
-	float resample_factor = resamp_pvt->resample_factor;
-
-	return resample_frame(pvt, resampler, resample_factor, f);
-}
-
-static int slin8_to_slin16_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
-{
-	struct slin8_to_slin16_pvt *resamp_pvt = pvt->pvt;
-	void *resampler = resamp_pvt->resampler;
-	float resample_factor = resamp_pvt->resample_factor;
-
-	return resample_frame(pvt, resampler, resample_factor, f);
+	return 0;
 }
 
-static struct ast_translator slin16_to_slin8 = {
-	.name = "slin16_to_slin8",
-	.newpvt = slin16_to_slin8_new,
-	.destroy = slin16_to_slin8_destroy,
-	.framein = slin16_to_slin8_framein,
-	.sample = slin16_sample,
-	.desc_size = sizeof(struct slin16_to_slin8_pvt),
-	.buffer_samples = (OUTBUF_SIZE / sizeof(int16_t)),
-	.buf_size = OUTBUF_SIZE,
-};
-
-static struct ast_translator slin8_to_slin16 = {
-	.name = "slin8_to_slin16",
-	.newpvt = slin8_to_slin16_new,
-	.destroy = slin8_to_slin16_destroy,
-	.framein = slin8_to_slin16_framein,
-	.sample = slin8_sample,
-	.desc_size = sizeof(struct slin8_to_slin16_pvt),
-	.buffer_samples = (OUTBUF_SIZE / sizeof(int16_t)),
-	.buf_size = OUTBUF_SIZE,
-};
-
 static int unload_module(void)
 {
 	int res = 0;
+	int idx;
 
-	res |= ast_unregister_translator(&slin16_to_slin8);
-	res |= ast_unregister_translator(&slin8_to_slin16);
+	for (idx = 0; idx < trans_size; idx++) {
+		res |= ast_unregister_translator(&translators[idx]);
+	}
+	ast_free(translators);
 
 	return res;
 }
@@ -203,15 +106,33 @@ static int unload_module(void)
 static int load_module(void)
 {
 	int res = 0;
+	int x, y, idx = 0;
 
-	ast_format_set(&slin16_to_slin8.src_format, AST_FORMAT_SLINEAR16, 0);
-	ast_format_set(&slin16_to_slin8.dst_format, AST_FORMAT_SLINEAR, 0);
+	trans_size = ARRAY_LEN(id_list) * ARRAY_LEN(id_list);
+	if (!(translators = ast_calloc(1, sizeof(struct ast_translator) * trans_size))) {
+		return AST_MODULE_LOAD_FAILURE;
+	}
 
-	ast_format_set(&slin8_to_slin16.src_format, AST_FORMAT_SLINEAR, 0);
-	ast_format_set(&slin8_to_slin16.dst_format, AST_FORMAT_SLINEAR16, 0);
+	for (x = 0; x < ARRAY_LEN(id_list); x++) {
+		for (y = 0; y < ARRAY_LEN(id_list); y++) {
+			if (x == y) {
+				continue;
+			}
+			translators[idx].newpvt = resamp_new;
+			translators[idx].destroy = resamp_destroy;
+			translators[idx].framein = resamp_framein;
+			translators[idx].desc_size = 0;
+			translators[idx].buffer_samples = (OUTBUF_SIZE / sizeof(int16_t));
+			translators[idx].buf_size = OUTBUF_SIZE;
+			ast_format_set(&translators[idx].src_format, id_list[x], 0);
+			ast_format_set(&translators[idx].dst_format, id_list[y], 0);
+			snprintf(translators[idx].name, sizeof(translators[idx].name), "slin %dkhz -> %dkhz",
+				ast_format_rate(&translators[idx].src_format), ast_format_rate(&translators[idx].dst_format));
+			res |= ast_register_translator(&translators[idx]);
+			idx++;
+		}
 
-	res |= ast_register_translator(&slin16_to_slin8);
-	res |= ast_register_translator(&slin8_to_slin16);
+	}
 
 	return AST_MODULE_LOAD_SUCCESS;
 }