Update Chromium to branch 1650 (31.0.1650.63)

Change-Id: I57d8c832eaec1eb2364e0a8e7352a6dd354db99f
Reviewed-by: Jocelyn Turcotte <jocelyn.turcotte@digia.com>

1 files changed, 55 insertions, 153 deletions

diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
index f35ec04c2fc..f5f84505fe3 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
@@ -17,7 +17,7 @@
 
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/modules/audio_coding/main/source/acm_dtmf_detection.h"
 #include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
 #include "webrtc/modules/audio_coding/main/source/acm_resampler.h"
@@ -32,6 +32,8 @@
 
 namespace webrtc {
 
+namespace acm1 {
+
 enum {
   kACMToneEnd = 999
 };
@@ -574,7 +576,7 @@ int AudioCodingModuleImpl::ProcessSingleStream() {
         }
         case kActiveNormalEncoded:
         case kPassiveNormalEncoded: {
-          current_payload_type = (uint8_t) send_codec_inst_.pltype;
+          current_payload_type = static_cast<uint8_t>(send_codec_inst_.pltype);
           frame_type = kAudioFrameSpeech;
           break;
         }
@@ -662,8 +664,8 @@ int AudioCodingModuleImpl::ProcessSingleStream() {
                  fragmentation_.fragmentationLength[1]);
           // Update the fragmentation time difference vector, in number of
           // timestamps.
-          uint16_t time_since_last = uint16_t(
-              rtp_timestamp - last_fec_timestamp_);
+          uint16_t time_since_last = static_cast<uint16_t>(rtp_timestamp -
+                                                           last_fec_timestamp_);
 
           // Update fragmentation vectors.
           fragmentation_.fragmentationPlType[1] =
@@ -736,7 +738,7 @@ int AudioCodingModuleImpl::ProcessSingleStream() {
 
     if (vad_callback_ != NULL) {
       // Callback with VAD decision.
-      vad_callback_->InFrameType(((int16_t) encoding_type));
+      vad_callback_->InFrameType(static_cast<int16_t>(encoding_type));
     }
   }
   return length_bytes;
@@ -1060,7 +1062,6 @@ int32_t AudioCodingModuleImpl::RegisterSendCodec(
     }
 
     ACMGenericCodec* codec_ptr = codecs_[codec_id];
-    int16_t status;
     WebRtcACMCodecParams codec_params;
 
     memcpy(&(codec_params.codec_inst), &send_codec, sizeof(CodecInst));
@@ -1068,12 +1069,7 @@ int32_t AudioCodingModuleImpl::RegisterSendCodec(
     codec_params.enable_dtx = dtx_enabled_;
     codec_params.vad_mode = vad_mode_;
     // Force initialization.
-    status = codec_ptr->InitEncoder(&codec_params, true);
-
-    // Check if VAD was turned on, or if error is reported.
-    if (status == 1) {
-      vad_enabled_ = true;
-    } else if (status < 0) {
+    if (codec_ptr->InitEncoder(&codec_params, true) < 0) {
       // Could not initialize the encoder.
 
       // Check if already have a registered codec.
@@ -1090,17 +1086,18 @@ int32_t AudioCodingModuleImpl::RegisterSendCodec(
       return -1;
     }
 
+    // Update states.
+    dtx_enabled_ = codec_params.enable_dtx;
+    vad_enabled_ = codec_params.enable_vad;
+    vad_mode_ = codec_params.vad_mode;
+
     // Everything is fine so we can replace the previous codec with this one.
     if (send_codec_registered_) {
       // If we change codec we start fresh with FEC.
       // This is not strictly required by the standard.
       is_first_red_ = true;
 
-      if (codec_ptr->SetVAD(dtx_enabled_, vad_enabled_, vad_mode_) < 0) {
-        // SetVAD failed.
-        vad_enabled_ = false;
-        dtx_enabled_ = false;
-      }
+      codec_ptr->SetVAD(&dtx_enabled_, &vad_enabled_, &vad_mode_);
     }
 
     current_send_codec_idx_ = codec_id;
@@ -1265,64 +1262,6 @@ int32_t AudioCodingModuleImpl::RegisterTransportCallback(
   return 0;
 }
 
-// Used by the module to deliver messages to the codec module/application
-// AVT(DTMF).
-int32_t AudioCodingModuleImpl::RegisterIncomingMessagesCallback(
-#ifndef WEBRTC_DTMF_DETECTION
-    AudioCodingFeedback* /* incoming_message */,
-    const ACMCountries /* cpt */) {
-  return -1;
-#else
-    AudioCodingFeedback* incoming_message,
-    const ACMCountries cpt) {
-  int16_t status = 0;
-
-  // Enter the critical section for callback.
-  {
-    CriticalSectionScoped lock(callback_crit_sect_);
-    dtmf_callback_ = incoming_message;
-  }
-  // Enter the ACM critical section to set up the DTMF class.
-  {
-    CriticalSectionScoped lock(acm_crit_sect_);
-    // Check if the call is to disable or enable the callback.
-    if (incoming_message == NULL) {
-      // Callback is disabled, delete DTMF-detector class.
-      if (dtmf_detector_ != NULL) {
-        delete dtmf_detector_;
-        dtmf_detector_ = NULL;
-      }
-      status = 0;
-    } else {
-      status = 0;
-      if (dtmf_detector_ == NULL) {
-        dtmf_detector_ = new ACMDTMFDetection;
-        if (dtmf_detector_ == NULL) {
-          status = -1;
-        }
-      }
-      if (status >= 0) {
-        status = dtmf_detector_->Enable(cpt);
-        if (status < 0) {
-          // Failed to initialize if DTMF-detection was not enabled before,
-          // delete the class, and set the callback to NULL and return -1.
-          delete dtmf_detector_;
-          dtmf_detector_ = NULL;
-        }
-      }
-    }
-  }
-  // Check if we failed in setting up the DTMF-detector class.
-  if ((status < 0)) {
-    // We failed, we cannot have the callback.
-    CriticalSectionScoped lock(callback_crit_sect_);
-    dtmf_callback_ = NULL;
-  }
-
-  return status;
-#endif
-}
-
 // Add 10MS of raw (PCM) audio data to the encoder.
 int32_t AudioCodingModuleImpl::Add10MsData(
     const AudioFrame& audio_frame) {
@@ -1436,8 +1375,8 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
   assert((secondary_encoder_.get() != NULL) ?
       secondary_send_codec_inst_.plfreq == send_codec_inst_.plfreq : true);
 
-  bool resample = ((int32_t) in_frame.sample_rate_hz_
-      != send_codec_inst_.plfreq);
+  bool resample = static_cast<int32_t>(in_frame.sample_rate_hz_) !=
+      send_codec_inst_.plfreq;
 
   // This variable is true if primary codec and secondary codec (if exists)
   // are both mono and input is stereo.
@@ -1489,8 +1428,8 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
     // Calculate the timestamp of this frame.
     if (last_in_timestamp_ > in_frame.timestamp_) {
       // A wrap around has happened.
-      timestamp_diff = ((uint32_t) 0xFFFFFFFF - last_in_timestamp_)
-                                                 + in_frame.timestamp_;
+      timestamp_diff = (static_cast<uint32_t>(0xFFFFFFFF) - last_in_timestamp_)
+          + in_frame.timestamp_;
     } else {
       timestamp_diff = in_frame.timestamp_ - last_in_timestamp_;
     }
@@ -1556,15 +1495,13 @@ AudioCodingModuleImpl::SetFECStatus(
 /////////////////////////////////////////
 //   (VAD) Voice Activity Detection
 //
-int32_t AudioCodingModuleImpl::SetVAD(const bool enable_dtx,
-                                      const bool enable_vad,
-                                      const ACMVADMode mode) {
+int32_t AudioCodingModuleImpl::SetVAD(bool enable_dtx, bool enable_vad,
+                                      ACMVADMode mode) {
   CriticalSectionScoped lock(acm_crit_sect_);
   return SetVADSafe(enable_dtx, enable_vad, mode);
 }
 
-int AudioCodingModuleImpl::SetVADSafe(bool enable_dtx,
-                                      bool enable_vad,
+int AudioCodingModuleImpl::SetVADSafe(bool enable_dtx, bool enable_vad,
                                       ACMVADMode mode) {
   // Sanity check of the mode.
   if ((mode != VADNormal) && (mode != VADLowBitrate)
@@ -1579,7 +1516,10 @@ int AudioCodingModuleImpl::SetVADSafe(bool enable_dtx,
   // sending.
   if ((enable_dtx || enable_vad) && stereo_send_) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
-                 "VAD/DTX not supported for stereo sending");
+                 "VAD/DTX not supported for stereo sending.");
+    dtx_enabled_ = false;
+    vad_enabled_ = false;
+    vad_mode_ = mode;
     return -1;
   }
 
@@ -1588,37 +1528,31 @@ int AudioCodingModuleImpl::SetVADSafe(bool enable_dtx,
   if ((enable_dtx || enable_vad) && secondary_encoder_.get() != NULL) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                  "VAD/DTX not supported when dual-streaming is enabled.");
+    dtx_enabled_ = false;
+    vad_enabled_ = false;
+    vad_mode_ = mode;
     return -1;
   }
 
+  // Store VAD/DTX settings. Values can be changed in the call to "SetVAD"
+  // below.
+  dtx_enabled_ = enable_dtx;
+  vad_enabled_ = enable_vad;
+  vad_mode_ = mode;
+
   // If a send codec is registered, set VAD/DTX for the codec.
   if (HaveValidEncoder("SetVAD")) {
-    int16_t status = codecs_[current_send_codec_idx_]->SetVAD(enable_dtx,
-                                                              enable_vad,
-                                                              mode);
-    if (status == 1) {
-      // Vad was enabled.
-      vad_enabled_ = true;
-      dtx_enabled_ = enable_dtx;
-      vad_mode_ = mode;
-
-      return 0;
-    } else if (status < 0) {
+    if (codecs_[current_send_codec_idx_]->SetVAD(&dtx_enabled_, &vad_enabled_,
+                                                 &vad_mode_) < 0) {
       // SetVAD failed.
       WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                    "SetVAD failed");
-
-      vad_enabled_ = false;
       dtx_enabled_ = false;
-
+      vad_enabled_ = false;
       return -1;
     }
   }
 
-  vad_enabled_ = enable_vad;
-  dtx_enabled_ = enable_dtx;
-  vad_mode_ = mode;
-
   return 0;
 }
 
@@ -2205,13 +2139,7 @@ int AudioCodingModuleImpl::InitStereoSlave() {
   return 0;
 }
 
-// Minimum playout delay (Used for lip-sync).
 int AudioCodingModuleImpl::SetMinimumPlayoutDelay(int time_ms) {
-  if ((time_ms < 0) || (time_ms > 10000)) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
-                 "Delay must be in the range of 0-10000 milliseconds.");
-    return -1;
-  }
   {
     CriticalSectionScoped lock(acm_crit_sect_);
     // Don't let the extra delay modified while accumulating buffers in NetEq.
@@ -2221,6 +2149,10 @@ int AudioCodingModuleImpl::SetMinimumPlayoutDelay(int time_ms) {
   return neteq_.SetMinimumDelay(time_ms);
 }
 
+int AudioCodingModuleImpl::SetMaximumPlayoutDelay(int time_ms) {
+  return neteq_.SetMaximumDelay(time_ms);
+}
+
 // Get Dtmf playout status.
 bool AudioCodingModuleImpl::DtmfPlayoutStatus() const {
 #ifndef WEBRTC_CODEC_AVT
@@ -2381,7 +2313,7 @@ int32_t AudioCodingModuleImpl::PlayoutData10Ms(
       }
 
       // Set the payload data length from the resampler.
-      audio_frame->samples_per_channel_ = (uint16_t) temp_len;
+      audio_frame->samples_per_channel_ = static_cast<uint16_t>(temp_len);
       // Set the sampling frequency.
       audio_frame->sample_rate_hz_ = desired_freq_hz;
     } else {
@@ -2455,11 +2387,12 @@ int32_t AudioCodingModuleImpl::PlayoutData10Ms(
     if (dtmf_callback_ != NULL) {
       if (tone != kACMToneEnd) {
         // just a tone
-        dtmf_callback_->IncomingDtmf((uint8_t) tone, false);
+        dtmf_callback_->IncomingDtmf(static_cast<uint8_t>(tone), false);
       } else if ((tone == kACMToneEnd) && (last_detected_tone != kACMToneEnd)) {
         // The tone is "END" and the previously detected tone is
         // not "END," so call fir an end.
-        dtmf_callback_->IncomingDtmf((uint8_t) last_detected_tone, true);
+        dtmf_callback_->IncomingDtmf(static_cast<uint8_t>(last_detected_tone),
+                                     true);
       }
     }
   }
@@ -2472,26 +2405,11 @@ int32_t AudioCodingModuleImpl::PlayoutData10Ms(
 }
 
 /////////////////////////////////////////
-//   (CNG) Comfort Noise Generation
-//   Generate comfort noise when receiving DTX packets
-//
-
-// Get VAD aggressiveness on the incoming stream
-ACMVADMode AudioCodingModuleImpl::ReceiveVADMode() const {
-  return neteq_.vad_mode();
-}
-
-// Configure VAD aggressiveness on the incoming stream.
-int16_t AudioCodingModuleImpl::SetReceiveVADMode(const ACMVADMode mode) {
-  return neteq_.SetVADMode(mode);
-}
-
-/////////////////////////////////////////
 //   Statistics
 //
 
 int32_t AudioCodingModuleImpl::NetworkStatistics(
-    ACMNetworkStatistics* statistics) const {
+    ACMNetworkStatistics* statistics) {
   int32_t status;
   status = neteq_.NetworkStatistics(statistics);
   return status;
@@ -2660,7 +2578,7 @@ int16_t AudioCodingModuleImpl::DecoderListIDByPlName(
         assert(registered_pltypes_[id] >= 0);
         assert(registered_pltypes_[id] <= 255);
         codecs_[id]->DecoderParams(
-            &codec_params, (uint8_t) registered_pltypes_[id]);
+            &codec_params, static_cast<uint8_t>(registered_pltypes_[id]));
         if (!STR_CASE_CMP(codec_params.codec_inst.plname, name)) {
           // Check if the given sampling frequency matches.
           // A zero sampling frequency means we matching the names
@@ -2731,8 +2649,7 @@ int32_t AudioCodingModuleImpl::IsInternalDTXReplacedWithWebRtc(
   return 0;
 }
 
-int32_t AudioCodingModuleImpl::SetISACMaxRate(
-    const uint32_t max_bit_per_sec) {
+int AudioCodingModuleImpl::SetISACMaxRate(int max_bit_per_sec) {
   CriticalSectionScoped lock(acm_crit_sect_);
 
   if (!HaveValidEncoder("SetISACMaxRate")) {
@@ -2742,8 +2659,7 @@ int32_t AudioCodingModuleImpl::SetISACMaxRate(
   return codecs_[current_send_codec_idx_]->SetISACMaxRate(max_bit_per_sec);
 }
 
-int32_t AudioCodingModuleImpl::SetISACMaxPayloadSize(
-    const uint16_t max_size_bytes) {
+int AudioCodingModuleImpl::SetISACMaxPayloadSize(int max_size_bytes) {
   CriticalSectionScoped lock(acm_crit_sect_);
 
   if (!HaveValidEncoder("SetISACMaxPayloadSize")) {
@@ -2755,9 +2671,9 @@ int32_t AudioCodingModuleImpl::SetISACMaxPayloadSize(
 }
 
 int32_t AudioCodingModuleImpl::ConfigISACBandwidthEstimator(
-    const uint8_t frame_size_ms,
-    const uint16_t rate_bit_per_sec,
-    const bool enforce_frame_size) {
+    int frame_size_ms,
+    int rate_bit_per_sec,
+    bool enforce_frame_size) {
   CriticalSectionScoped lock(acm_crit_sect_);
 
   if (!HaveValidEncoder("ConfigISACBandwidthEstimator")) {
@@ -2768,21 +2684,6 @@ int32_t AudioCodingModuleImpl::ConfigISACBandwidthEstimator(
       frame_size_ms, rate_bit_per_sec, enforce_frame_size);
 }
 
-int32_t AudioCodingModuleImpl::SetBackgroundNoiseMode(
-    const ACMBackgroundNoiseMode mode) {
-  if ((mode < On) || (mode > Off)) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
-                 "The specified background noise is out of range.\n");
-    return -1;
-  }
-  return neteq_.SetBackgroundNoiseMode(mode);
-}
-
-int32_t AudioCodingModuleImpl::BackgroundNoiseMode(
-    ACMBackgroundNoiseMode* mode) {
-  return neteq_.BackgroundNoiseMode(*mode);
-}
-
 int32_t AudioCodingModuleImpl::PlayoutTimestamp(
     uint32_t* timestamp) {
   WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
@@ -2818,8 +2719,7 @@ bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const {
   return true;
 }
 
-int32_t AudioCodingModuleImpl::UnregisterReceiveCodec(
-    const int16_t payload_type) {
+int AudioCodingModuleImpl::UnregisterReceiveCodec(uint8_t payload_type) {
   CriticalSectionScoped lock(acm_crit_sect_);
   int id;
 
@@ -3126,4 +3026,6 @@ void AudioCodingModuleImpl::DisableNack() {
   nack_enabled_ = false;
 }
 
+}  // namespace acm1
+
 }  // namespace webrtc