Update a ton of audio code to use size_t more correctly and in general reduce

webrtc/modules/audio_coding/codecs/opus/opus_interface.c

Index: webrtc/modules/audio_coding/codecs/opus/opus_interface.c
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
index e2506166a923a07b3599c5d3728f0ae40328facd..e2a8383c4b665da3982843be94a360946d4f8d49 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
@@ -80,8 +80,8 @@ int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) {
 
 int WebRtcOpus_Encode(OpusEncInst* inst,
                       const int16_t* audio_in,
-                      int16_t samples,
-                      int16_t length_encoded_buffer,
+                      size_t samples,
+                      size_t length_encoded_buffer,
                       uint8_t* encoded) {
   int res;
 
@@ -91,9 +91,9 @@ int WebRtcOpus_Encode(OpusEncInst* inst,
 
   res = opus_encode(inst->encoder,
                     (const opus_int16*)audio_in,
-                    samples,
+                    (int)samples,
                     encoded,
-                    length_encoded_buffer);
+                    (opus_int32)length_encoded_buffer);
 
   if (res == 1) {
     // Indicates DTX since the packet has nothing but a header. In principle,
@@ -260,7 +260,7 @@ int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst) {
 }
 
 /* For decoder to determine if it is to output speech or comfort noise. */
-static int16_t DetermineAudioType(OpusDecInst* inst, int16_t encoded_bytes) {
+static int16_t DetermineAudioType(OpusDecInst* inst, size_t encoded_bytes) {
   // Audio type becomes comfort noise if |encoded_byte| is 1 and keeps
   // to be so if the following |encoded_byte| are 0 or 1.
   if (encoded_bytes == 0 && inst->in_dtx_mode) {
@@ -278,9 +278,9 @@ static int16_t DetermineAudioType(OpusDecInst* inst, int16_t encoded_bytes) {
  * is set to the number of samples needed for PLC in case of losses.
  * It is up to the caller to make sure the value is correct. */
 static int DecodeNative(OpusDecInst* inst, const uint8_t* encoded,
-                        int16_t encoded_bytes, int frame_size,
+                        size_t encoded_bytes, int frame_size,
                         int16_t* decoded, int16_t* audio_type, int decode_fec) {
-  int res = opus_decode(inst->decoder, encoded, encoded_bytes,
+  int res = opus_decode(inst->decoder, encoded, (opus_int32)encoded_bytes,
                         (opus_int16*)decoded, frame_size, decode_fec);
 
   if (res <= 0)
@@ -292,7 +292,7 @@ static int DecodeNative(OpusDecInst* inst, const uint8_t* encoded,
 }
 
 int WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
-                      int16_t encoded_bytes, int16_t* decoded,
+                      size_t encoded_bytes, int16_t* decoded,
                       int16_t* audio_type) {
   int decoded_samples;
 
@@ -340,7 +340,7 @@ int WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
 }
 
 int WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
-                         int16_t encoded_bytes, int16_t* decoded,
+                         size_t encoded_bytes, int16_t* decoded,
                          int16_t* audio_type) {
   int decoded_samples;
   int fec_samples;
@@ -362,9 +362,9 @@ int WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
 
 int WebRtcOpus_DurationEst(OpusDecInst* inst,
                            const uint8_t* payload,
-                           int payload_length_bytes) {
+                           size_t payload_length_bytes) {
   int frames, samples;
-  frames = opus_packet_get_nb_frames(payload, payload_length_bytes);
+  frames = opus_packet_get_nb_frames(payload, (opus_int32)payload_length_bytes);
   if (frames < 0) {
     /* Invalid payload data. */
     return 0;
@@ -378,7 +378,7 @@ int WebRtcOpus_DurationEst(OpusDecInst* inst,
 }
 
 int WebRtcOpus_FecDurationEst(const uint8_t* payload,
-                              int payload_length_bytes) {
+                              size_t payload_length_bytes) {
   int samples;
   if (WebRtcOpus_PacketHasFec(payload, payload_length_bytes) != 1) {
     return 0;
@@ -393,13 +393,13 @@ int WebRtcOpus_FecDurationEst(const uint8_t* payload,
 }
 
 int WebRtcOpus_PacketHasFec(const uint8_t* payload,
-                            int payload_length_bytes) {
+                            size_t payload_length_bytes) {
   int frames, channels, payload_length_ms;
   int n;
   opus_int16 frame_sizes[48];
   const unsigned char *frame_data[48];
 
-  if (payload == NULL || payload_length_bytes <= 0)
+  if (payload == NULL || payload_length_bytes == 0)
     return 0;
 
   /* In CELT_ONLY mode, packets should not have FEC. */
@@ -432,8 +432,8 @@ int WebRtcOpus_PacketHasFec(const uint8_t* payload,
   }
 
   /* The following is to parse the LBRR flags. */
-  if (opus_packet_parse(payload, payload_length_bytes, NULL, frame_data,
-                        frame_sizes, NULL) < 0) {
+  if (opus_packet_parse(payload, (opus_int32)payload_length_bytes, NULL,
+                        frame_data, frame_sizes, NULL) < 0) {
     return 0;
   }