1 files changed, 99 insertions, 0 deletions
diff --git a/src/audio/audio_codecs.cpp b/src/audio/audio_codecs.cpp
new file mode 100644
index 0000000..75a9262
--- /dev/null
+++ b/src/audio/audio_codecs.cpp
@@ -0,0 +1,99 @@
+/*
+    Copyright (C) 2005-2009  Michel de Boer <michel@twinklephone.com>
+
+    This program is free software; you can redistribute it and/or modify
+    it under the terms of the GNU General Public License as published by
+    the Free Software Foundation; either version 2 of the License, or
+    (at your option) any later version.
+
+    This program is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+    GNU General Public License for more details.
+
+    You should have received a copy of the GNU General Public License
+    along with this program; if not, write to the Free Software
+    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+*/
+
+#include <cstdlib>
+#include "audio_codecs.h"
+
+unsigned short audio_sample_rate(t_audio_codec codec) {
+	switch(codec) {
+	case CODEC_G711_ALAW:
+	case CODEC_G711_ULAW:
+	case CODEC_GSM:
+	case CODEC_SPEEX_NB:
+	case CODEC_ILBC:
+	case CODEC_G726_16:
+	case CODEC_G726_24:
+	case CODEC_G726_32:
+	case CODEC_G726_40:
+	case CODEC_TELEPHONE_EVENT:
+		return 8000;
+	case CODEC_SPEEX_WB:
+		return 16000;
+	case CODEC_SPEEX_UWB:
+		return 32000;
+	default:
+		// Use 8000 as default rate
+		return 8000;
+	}
+}
+
+bool is_speex_codec(t_audio_codec codec) {
+	return (codec == CODEC_SPEEX_NB ||
+		codec == CODEC_SPEEX_WB ||
+		codec == CODEC_SPEEX_UWB);
+}
+
+int resample(short *input_buf, int input_len, int input_sample_rate,
+	short *output_buf, int output_len, int output_sample_rate)
+{
+	if (input_sample_rate > output_sample_rate) {
+		int downsample_factor = input_sample_rate / output_sample_rate;
+		int output_idx = -1;
+		for (int i = 0; i < input_len; i += downsample_factor) {
+			output_idx = i / downsample_factor;
+			if (output_idx >= output_len) {
+				// Output buffer is full
+				return output_len;
+			}
+			output_buf[output_idx] = input_buf[i];
+		}
+		return output_idx + 1;
+	} else {
+		int upsample_factor = output_sample_rate / input_sample_rate;
+		int output_idx = -1;
+		for (int i = 0; i < input_len; i++) {
+			for (int j = 0; j < upsample_factor; j++) {
+				output_idx = i * upsample_factor + j;
+				if (output_idx >= output_len) {
+					// Output buffer is full
+					return output_len;
+				}
+				output_buf[output_idx] = input_buf[i];
+			}
+		}
+		return output_idx + 1;
+	}
+}
+
+short mix_linear_pcm(short pcm1, short pcm2) {
+	long mixed_sample = long(pcm1) + long(pcm2);
+
+	// Compress a 17 bit PCM value into a 16-bit value.
+	// The easy way is to divide the value by 2, but this lowers
+	// the volume.
+	// Only lower the volume for the loud values. As for a normal
+	// voice call the values are not that loud, this gives better
+	// quality.
+	if (mixed_sample > 16384) {
+		mixed_sample = 16384 + (mixed_sample - 16384) / 3;
+	} else if (mixed_sample < -16384) {
+		mixed_sample = -16384 - (-16384 - mixed_sample) / 3;
+	}
+
+	return short(mixed_sample);
+}