Convert channel counts to size_t.

IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.

BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1316523002 .

Cr-Commit-Position: refs/heads/master@{#11229}

5 files changed, 9 insertions, 9 deletions

diff --git a/talk/app/webrtc/mediastreaminterface.h b/talk/app/webrtc/mediastreaminterface.h
index 89a4abe7fc..9b137d9f76 100644
--- a/talk/app/webrtc/mediastreaminterface.h
+++ b/talk/app/webrtc/mediastreaminterface.h
@@ -151,7 +151,7 @@ class AudioTrackSinkInterface {
   virtual void OnData(const void* audio_data,
                       int bits_per_sample,
                       int sample_rate,
-                      int number_of_channels,
+                      size_t number_of_channels,
                       size_t number_of_frames) = 0;
 
  protected:
diff --git a/talk/app/webrtc/rtpsender.cc b/talk/app/webrtc/rtpsender.cc
index c0d23a0503..91e484b733 100644
--- a/talk/app/webrtc/rtpsender.cc
+++ b/talk/app/webrtc/rtpsender.cc
@@ -44,7 +44,7 @@ LocalAudioSinkAdapter::~LocalAudioSinkAdapter() {
 void LocalAudioSinkAdapter::OnData(const void* audio_data,
                                    int bits_per_sample,
                                    int sample_rate,
-                                   int number_of_channels,
+                                   size_t number_of_channels,
                                    size_t number_of_frames) {
   rtc::CritScope lock(&lock_);
   if (sink_) {
diff --git a/talk/app/webrtc/rtpsender.h b/talk/app/webrtc/rtpsender.h
index d5f88a941a..dd846b556c 100644
--- a/talk/app/webrtc/rtpsender.h
+++ b/talk/app/webrtc/rtpsender.h
@@ -57,7 +57,7 @@ class LocalAudioSinkAdapter : public AudioTrackSinkInterface,
   void OnData(const void* audio_data,
               int bits_per_sample,
               int sample_rate,
-              int number_of_channels,
+              size_t number_of_channels,
               size_t number_of_frames) override;
 
   // cricket::AudioRenderer implementation.
diff --git a/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc b/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
index e2dc12375b..6b675a9395 100644
--- a/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
+++ b/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
@@ -58,7 +58,7 @@ class FakeAdmTest : public testing::Test,
   int32_t RecordedDataIsAvailable(const void* audioSamples,
                                   const size_t nSamples,
                                   const size_t nBytesPerSample,
-                                  const uint8_t nChannels,
+                                  const size_t nChannels,
                                   const uint32_t samplesPerSec,
                                   const uint32_t totalDelayMS,
                                   const int32_t clockDrift,
@@ -82,7 +82,7 @@ class FakeAdmTest : public testing::Test,
   // ADM is pulling data.
   int32_t NeedMorePlayData(const size_t nSamples,
                            const size_t nBytesPerSample,
-                           const uint8_t nChannels,
+                           const size_t nChannels,
                            const uint32_t samplesPerSec,
                            void* audioSamples,
                            size_t& nSamplesOut,
diff --git a/talk/app/webrtc/webrtcsdp.cc b/talk/app/webrtc/webrtcsdp.cc
index 07a4eb92f4..e287e90916 100644
--- a/talk/app/webrtc/webrtcsdp.cc
+++ b/talk/app/webrtc/webrtcsdp.cc
@@ -2064,7 +2064,7 @@ static bool ParseDtlsSetup(const std::string& line,
 struct StaticPayloadAudioCodec {
   const char* name;
   int clockrate;
-  int channels;
+  size_t channels;
 };
 static const StaticPayloadAudioCodec kStaticPayloadAudioCodecs[] = {
   { "PCMU", 8000, 1 },
@@ -2103,7 +2103,7 @@ void MaybeCreateStaticPayloadAudioCodecs(
         payload_type < arraysize(kStaticPayloadAudioCodecs)) {
       std::string encoding_name = kStaticPayloadAudioCodecs[payload_type].name;
       int clock_rate = kStaticPayloadAudioCodecs[payload_type].clockrate;
-      int channels = kStaticPayloadAudioCodecs[payload_type].channels;
+      size_t channels = kStaticPayloadAudioCodecs[payload_type].channels;
       media_desc->AddCodec(cricket::AudioCodec(payload_type, encoding_name,
                                                clock_rate, 0, channels,
                                                preference));
@@ -2838,7 +2838,7 @@ bool ParseCryptoAttribute(const std::string& line,
 // Updates or creates a new codec entry in the audio description with according
 // to |name|, |clockrate|, |bitrate|, |channels| and |preference|.
 void UpdateCodec(int payload_type, const std::string& name, int clockrate,
-                 int bitrate, int channels, int preference,
+                 int bitrate, size_t channels, int preference,
                  AudioContentDescription* audio_desc) {
   // Codec may already be populated with (only) optional parameters
   // (from an fmtp).
@@ -2937,7 +2937,7 @@ bool ParseRtpmapAttribute(const std::string& line,
     // of audio channels.  This parameter is OPTIONAL and may be
     // omitted if the number of channels is one, provided that no
     // additional parameters are needed.
-    int channels = 1;
+    size_t channels = 1;
     if (codec_params.size() == 3) {
       if (!GetValueFromString(line, codec_params[2], &channels, error)) {
         return false;