Move talk/app/webrtc to webrtc/api

talk/app/webrtc/dtmfsender.cc

Index: talk/app/webrtc/dtmfsender.cc
diff --git a/talk/app/webrtc/dtmfsender.cc b/talk/app/webrtc/dtmfsender.cc
deleted file mode 100644
index 30e2ce3873deca75717e3a0a2037caf6fd4a88e3..0000000000000000000000000000000000000000
--- a/talk/app/webrtc/dtmfsender.cc
+++ /dev/null
@@ -1,255 +0,0 @@
-/*
- * libjingle
- * Copyright 2012 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "talk/app/webrtc/dtmfsender.h"
-
-#include <ctype.h>
-
-#include <string>
-
-#include "webrtc/base/logging.h"
-#include "webrtc/base/thread.h"
-
-namespace webrtc {
-
-enum {
-  MSG_DO_INSERT_DTMF = 0,
-};
-
-// RFC4733
-//  +-------+--------+------+---------+
-//  | Event | Code   | Type | Volume? |
-//  +-------+--------+------+---------+
-//  | 0--9  | 0--9   | tone | yes     |
-//  | *     | 10     | tone | yes     |
-//  | #     | 11     | tone | yes     |
-//  | A--D  | 12--15 | tone | yes     |
-//  +-------+--------+------+---------+
-// The "," is a special event defined by the WebRTC spec. It means to delay for
-// 2 seconds before processing the next tone. We use -1 as its code.
-static const int kDtmfCodeTwoSecondDelay = -1;
-static const int kDtmfTwoSecondInMs = 2000;
-static const char kDtmfValidTones[] = ",0123456789*#ABCDabcd";
-static const char kDtmfTonesTable[] = ",0123456789*#ABCD";
-// The duration cannot be more than 6000ms or less than 70ms. The gap between
-// tones must be at least 50 ms.
-static const int kDtmfDefaultDurationMs = 100;
-static const int kDtmfMinDurationMs = 70;
-static const int kDtmfMaxDurationMs = 6000;
-static const int kDtmfDefaultGapMs = 50;
-static const int kDtmfMinGapMs = 50;
-
-// Get DTMF code from the DTMF event character.
-bool GetDtmfCode(char tone, int* code) {
-  // Convert a-d to A-D.
-  char event = toupper(tone);
-  const char* p = strchr(kDtmfTonesTable, event);
-  if (!p) {
-    return false;
-  }
-  *code = p - kDtmfTonesTable - 1;
-  return true;
-}
-
-rtc::scoped_refptr<DtmfSender> DtmfSender::Create(
-    AudioTrackInterface* track,
-    rtc::Thread* signaling_thread,
-    DtmfProviderInterface* provider) {
-  if (!track || !signaling_thread) {
-    return NULL;
-  }
-  rtc::scoped_refptr<DtmfSender> dtmf_sender(
-      new rtc::RefCountedObject<DtmfSender>(track, signaling_thread,
-                                                  provider));
-  return dtmf_sender;
-}
-
-DtmfSender::DtmfSender(AudioTrackInterface* track,
-                       rtc::Thread* signaling_thread,
-                       DtmfProviderInterface* provider)
-    : track_(track),
-      observer_(NULL),
-      signaling_thread_(signaling_thread),
-      provider_(provider),
-      duration_(kDtmfDefaultDurationMs),
-      inter_tone_gap_(kDtmfDefaultGapMs) {
-  ASSERT(track_ != NULL);
-  ASSERT(signaling_thread_ != NULL);
-  // TODO(deadbeef): Once we can use shared_ptr and weak_ptr,
-  // do that instead of relying on a "destroyed" signal.
-  if (provider_) {
-    ASSERT(provider_->GetOnDestroyedSignal() != NULL);
-    provider_->GetOnDestroyedSignal()->connect(
-        this, &DtmfSender::OnProviderDestroyed);
-  }
-}
-
-DtmfSender::~DtmfSender() {
-  StopSending();
-}
-
-void DtmfSender::RegisterObserver(DtmfSenderObserverInterface* observer) {
-  observer_ = observer;
-}
-
-void DtmfSender::UnregisterObserver() {
-  observer_ = NULL;
-}
-
-bool DtmfSender::CanInsertDtmf() {
-  ASSERT(signaling_thread_->IsCurrent());
-  if (!provider_) {
-    return false;
-  }
-  return provider_->CanInsertDtmf(track_->id());
-}
-
-bool DtmfSender::InsertDtmf(const std::string& tones, int duration,
-                            int inter_tone_gap) {
-  ASSERT(signaling_thread_->IsCurrent());
-
-  if (duration > kDtmfMaxDurationMs ||
-      duration < kDtmfMinDurationMs ||
-      inter_tone_gap < kDtmfMinGapMs) {
-    LOG(LS_ERROR) << "InsertDtmf is called with invalid duration or tones gap. "
-        << "The duration cannot be more than " << kDtmfMaxDurationMs
-        << "ms or less than " << kDtmfMinDurationMs << "ms. "
-        << "The gap between tones must be at least " << kDtmfMinGapMs << "ms.";
-    return false;
-  }
-
-  if (!CanInsertDtmf()) {
-    LOG(LS_ERROR)
-        << "InsertDtmf is called on DtmfSender that can't send DTMF.";
-    return false;
-  }
-
-  tones_ = tones;
-  duration_ = duration;
-  inter_tone_gap_ = inter_tone_gap;
-  // Clear the previous queue.
-  signaling_thread_->Clear(this, MSG_DO_INSERT_DTMF);
-  // Kick off a new DTMF task queue.
-  signaling_thread_->Post(this, MSG_DO_INSERT_DTMF);
-  return true;
-}
-
-const AudioTrackInterface* DtmfSender::track() const {
-  return track_;
-}
-
-std::string DtmfSender::tones() const {
-  return tones_;
-}
-
-int DtmfSender::duration() const {
-  return duration_;
-}
-
-int DtmfSender::inter_tone_gap() const {
-  return inter_tone_gap_;
-}
-
-void DtmfSender::OnMessage(rtc::Message* msg) {
-  switch (msg->message_id) {
-    case MSG_DO_INSERT_DTMF: {
-      DoInsertDtmf();
-      break;
-    }
-    default: {
-      ASSERT(false);
-      break;
-    }
-  }
-}
-
-void DtmfSender::DoInsertDtmf() {
-  ASSERT(signaling_thread_->IsCurrent());
-
-  // Get the first DTMF tone from the tone buffer. Unrecognized characters will
-  // be ignored and skipped.
-  size_t first_tone_pos = tones_.find_first_of(kDtmfValidTones);
-  int code = 0;
-  if (first_tone_pos == std::string::npos) {
-    tones_.clear();
-    // Fire a “OnToneChange” event with an empty string and stop.
-    if (observer_) {
-      observer_->OnToneChange(std::string());
-    }
-    return;
-  } else {
-    char tone = tones_[first_tone_pos];
-    if (!GetDtmfCode(tone, &code)) {
-      // The find_first_of(kDtmfValidTones) should have guarantee |tone| is
-      // a valid DTMF tone.
-      ASSERT(false);
-    }
-  }
-
-  int tone_gap = inter_tone_gap_;
-  if (code == kDtmfCodeTwoSecondDelay) {
-    // Special case defined by WebRTC - The character',' indicates a delay of 2
-    // seconds before processing the next character in the tones parameter.
-    tone_gap = kDtmfTwoSecondInMs;
-  } else {
-    if (!provider_) {
-      LOG(LS_ERROR) << "The DtmfProvider has been destroyed.";
-      return;
-    }
-    // The provider starts playout of the given tone on the
-    // associated RTP media stream, using the appropriate codec.
-    if (!provider_->InsertDtmf(track_->id(), code, duration_)) {
-      LOG(LS_ERROR) << "The DtmfProvider can no longer send DTMF.";
-      return;
-    }
-    // Wait for the number of milliseconds specified by |duration_|.
-    tone_gap += duration_;
-  }
-
-  // Fire a “OnToneChange” event with the tone that's just processed.
-  if (observer_) {
-    observer_->OnToneChange(tones_.substr(first_tone_pos, 1));
-  }
-
-  // Erase the unrecognized characters plus the tone that's just processed.
-  tones_.erase(0, first_tone_pos + 1);
-
-  // Continue with the next tone.
-  signaling_thread_->PostDelayed(tone_gap, this, MSG_DO_INSERT_DTMF);
-}
-
-void DtmfSender::OnProviderDestroyed() {
-  LOG(LS_INFO) << "The Dtmf provider is deleted. Clear the sending queue.";
-  StopSending();
-  provider_ = NULL;
-}
-
-void DtmfSender::StopSending() {
-  signaling_thread_->Clear(this);
-}
-
-}  // namespace webrtc